Ipss4 Um

User Manual
IPSX4
ipss4_r2a
SIP VoIP Subscriber Media Gateway
MileGate
MileGate
IPSS4 User Manual
Copyright and Confidentiality Copyright in this document vests in KEYMILE. This document contains confi-
dential information which is the property of KEYMILE. It must be held in con-
fidence by the recipient and may not be used for any purposes except those
specifically authorised by contract or otherwise in writing by KEYMILE. This
document may not be copied in whole or in part, or any of its contents dis-
closed by the recipient to any third party, without the prior written agreement
of KEYMILE.
Disclaimer KEYMILE has taken reasonable care in compiling this document, however
KEYMILE accepts no liability whatsoever for any error or omission in the
information contained herein and gives no other warranty or undertaking as
to its accuracy.
KEYMILE reserves the right to amend this document at any time without
prior notice.
Document PEC EN/LZTBU 371 149 RA
Document release MileGate R4C | 23. July 2014
Published by KEYMILE
http://www.keymile.com
User Manual
IPSS4
Table of Content
1 Preface 7
1.1 Precautions and safety 7
1.2 Symbols and notations 7
1.3 Interfaces and circuit categories 7
1.4 Document history 8
2 Introduction 9
2.1 General 9
2.2 Unit view 11
2.3 Block diagram 12
3 Functions and specifications 16

3.1 Feature licences 16
3.2 Summary of standards 16
3.3 Main functions and specifications 19
3.4 Call progress tones 21
4 Installation 22
4.1 Prerequisites 22
4.2 Slots for the IPSS4 unit 22
4.3 Compatibility 23
4.4 Connections and cables 23
5 Functional Description 24
5.1 Applications 24
5.2 SIP protocol 28
5.3 Basic functions and protocols 36
5.4 Specific gateway (GW) functions 41
5.5 Proxy server and registrar server functions 77
© KEYMILE July 2014 page 3 of 207 EN/LZTBU 371 149 RA

User Manual
IPSS4
5.6 Direct dialling in and trunk hunting 86

5.7 Equipment protection (EQP) 88
5.8 Synchronization 92
5.9 Ethernet switch 92
6 Commissioning 93
6.1 Customer profiles 93
6.2 Profiles 95
6.3 Commissioning of a gateway 98
6.4 Gateway configuration with equipment protection 113
7 Operation 115
7.1 Port status management function 115
7.2 Unit optical indicators 117
7.3 Possible faults and related debugging 118
7.4 Maintenance 124
8 User interface reference 127

8.1 Introduction 127
8.2 Profiles 131
8.3 AP: / unit-x: IPSS4 139
8.4 AP: / unit-x / control 171
8.5 AP: / unit-x / media 172
8.6 AP: / unit-x / huntgroups 173
8.7 AP: / unit-x / huntgroups / group-a (ISDN-BA) 175
8.8 AP: / unit-x / portgroup-y (PSTN) or (ISDN-BA) 184
8.9 AP: / unit-x / portgroup-y / port-z (PSTN) 186
8.10 AP: / unit-x / portgroup-y / port-z (ISDN-BA) 194
8.11 AP: / unit-x / portgroup-y / icChannel 202
9 Annex 206
9.1 Associated MileGate documents 206

User Manual
IPSS4
9.2 Technical support 207

9.3 Product training 207

User Manual
IPSS4
Figures
Figure 1: Application example with IPSS4 9

Figure 2: IPSX4 unit view 11
Figure 3: IPSS4 functional block diagram for PSTN access 12
Figure 4: IPSS4 functional block diagram for ISDN-BA access 14
Figure 5: MileGate 2510 equipped for 912 PSTN subscribers with 15 SUPM5 units 25
Figure 6: MileGate 2310 and MileGate 2510 equipped for 304 ISDN-BA subscribers with 19 SUIx1 units
and one centralized IPSS4 unit 25
Figure 7: Example of the SIP connection model 28
Figure 8: SIP layers 29
Figure 9: SIP transactions 29
Figure 10: NGN network set up for the illustration of call establishment and call release 32
Figure 11: Call establishment with SIP 34
Figure 12: Call release with SIP 35
Figure 13: Basic function and protocol view for signalling and voice traffic in the case of PSTN access 36
Figure 14: Basic function and protocol view for signalling and voice traffic in the case of ISDN-BA access39
Figure 15: Functional blocks in the gateway 41
Figure 16: Basic view for fax transmission without T.38 fax relay 44
Figure 17: Basic view for fax transmission with T.38 fax relay 44
Figure 18: Block diagram, showing the T.38 fax relay function in the case of PSTN access 45
Figure 19: Protocol view for T.38 fax relay function in the case of PSTN access (assuming that V.17 is used
as fax transmission protocol) 45
Figure 20: Functional blocks of the echo canceller 47
Figure 21: Functional blocks for the silence suppression 48
Figure 22: Request retransmissions 53
Figure 23: Outgoing overlap signalling 56
Figure 24: Call release with expiry of the T10 timer 56
Figure 25: Incoming overlap signalling 58
Figure 26: Call scenario with three participants 60
Figure 27: Digit collection example 76
Figure 28: Parameter assignment from profiles and from the unit configuration 97
Figure 29: Subscriber number to port assignment in MileGate 98
Figure 30: Set-up for the implementation of the NGN voice service 100
Figure 31: Fault indication LEDs on the IPSS4 unit 117
Figure 32: MOM (managed object model) of the IPSS4 unit 128

Preface User Manual
IPSS4
1 Preface
1.1 Precautions and safety
Before you handle any equipment you must comply with the safety advices.
Adherence to the safety instructions ensures compliance with the safety
requirements as defined in EN 60950 (Safety of Information Technology
Equipment).
Please refer to the following document:
[202] Safety Instructions “Precautions and safety”.
1.2 Symbols and notations
This User Manual uses the following symbols:
Risk of operating trouble!

Indicates that an action may lead to operating trouble or loss of data.
→ Possible actions are given.
Please note:
Shows significant information.
→ Possible actions are given.
1.3 Interfaces and circuit categories
Table 1: Electrical interfaces and circuit categories

IPSS4 interface Circuit category according to Max. rating
EN 60950-1 Voltage Current
Local power supply TNV2 < 72 VDC < 1.0 A

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Preface User Manual
IPSS4
1.4 Document history
Table 2: Document history

KEYMILE PEC Date MileGate Changes since previous version
release
EN/LZTBU 371 149 RA June 2014 R4C This is the first release for the MileGate system release
R4C

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Introduction User Manual
IPSS4
2 Introduction
This section presents a general introduction to the IPSS4 unit.
The section is closed with a unit view in section 2.2 "Unit view" (on page 11)
and a block diagram overview in section 2.3 "Block diagram" (on page 12).
2.1 General
This document describes the architecture and functions of the IPSS4 unit
and shows, how this unit is commissioned and operated as part of the
MileGate.
The IPSS4 is a 1-slot wide service unit of MileGate. It acts as SIP gateway
for voice over IP for PSTN and ISDN-BA user ports.
The IPSS4 unit converts TDM based voice traffic into IP packets. It accesses
PSTN (POTS) and ISDN Basic-Rate Access (BA) services via PBUS from
the corresponding service units or from subtended network elements via a
TDM connection, e.g. with a LOMI8 unit.
The IPSS4 is a high capacity SIP gateway using the IPSX4 hardware unit. It
can serve up to 912 PSTN subscribers or up to 304 ISDN-BA subscribers
together with up to 304 PSTN subscribers.
The IP data signal is connected to the core unit COGE5 via the double GbE
star. The core unit offers the optical and electrical Ethernet interfaces
towards the IP network.
Softswitch Softswitch
Registrar Signalling Gateway SS7
Proxy SIP-H.248 MGC
SIP
MileGate Class 5
H.248/ Switch
PSTN SIP MEGACO
service unit
SIP
SUPMx access
10 GbE
Analogue Phone
core unit IP Trunking PSTN
gateway Network Gateway
ISDN-BA COGE5
service unit IPSS4
SUIx1 NGN voice traffic
ISDN Phone
Physical connection
Logical path for NGN voice
Logical path for NGN signalling
Figure 1: Application example with IPSS4
Please note:
ISDN-BA subscriber units will be available in a future release.
With the traditional voice application architecture, a switch or local exchange
contains both the signalling and the transport (switching fabric) functions.

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IPSS4
In a SIP Next Generation Network (NGN), the two functions signalling and
transport are both located in the gateway. The SIP softswitch functions
“proxy server” and “registrar server” just support the routing of the calls, they
are not mandatory. Accordingly, the following elements are introduced by the
SIP NGN architecture:
• The gateway (GW) is responsible for the media stream conversion, i.e.
the conversion of TDM based voice signals into IP packets and the sig-
nalling protocol termination. The GW supports line side interfaces, e.g. for
analogue or ISDN-BA phones.
• The SIP softswitch contains the SIP proxy server and SIP registrar server
functions. The registrar makes the location of a subscriber available to
the proxy server. The proxy forwards the SIP call control messages into
the network.
The SIP/MEGACO softswitch handles both the SIP and the H.248/MEG-
ACO protocols. This type of softswitch controls the trunking gateway and
implements also the signalling gateway to the SS7.

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IPSS4
2.2 Unit view
Figure 2: IPSX4 unit view
Figure 2 shows the IPSS4 unit hardware, named IPSX4. On the front plate
are two LEDs for unit- and traffic failure indication. No externally accessible
connectors are available.

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IPSS4
2.3 Block diagram
2.3.1 Implementation for PSTN access
Figure 3 shows the block diagram of the IPSS4 unit functions for PSTN
access.
PSTN signalling
Host processor
(out of band )
PSTN signalling NIF & user port management
RTCP handling
Signalling TDM / packet connection

processor manager SIP
PBUS access
Signalling
detection / UDP
generation IP
Media stream
Voice processor
TDM cross connect
Voice activity
Fax detection DTMF detector
relay Echo canceller Tone generator
FSK signalling
Voice codec
Subrack internal
Backplane access
RTP RTCP
communication
UDP
IP
Power RTP stream SIP
GbE star interface

GbE star
Ethernet
Figure 3: IPSS4 functional block diagram for PSTN access
Description of components:
• PBUS access
The PBUS access provides the access to the proprietary MileGate TDM
bus.
The IPSS4 has a connection capacity of 32 x 2 Mbit/s, providing access
for up to 912 PSTN subscribers.
• TDM cross-connect
Each TDM channel from the PBUS contains both voice traffic and signal-
ling data. The TDM cross connect separates the voice from the signalling
data. It directs the TDM voice traffic to the voice processor and the sig-
nalling data to the signalling processor.
• Signalling processor
The signalling processor terminates signalling channels, i.e. detects and
generates signalling messages which are received from and sent to the
appropriate linecard (e.g. SUPM1).
• Host processor
The host processor hosts the following functional blocks:

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IPSS4
− The board controller runs the ESW and controls the functions of the
IPSS4 unit. The ESW is stored directly in the flash memory of the unit.
At least two ESW versions can be stored on one unit.
− Initialization of the control association between the GW and the softs-
witch.
− PSTN signalling nodal interworking function (NIF):
Responsible for converting the (out of band) PSTN signalling mes-
sages received from the signalling processor to SIP commands and
vice versa for different national protocols.
− TDM ↔ Packet connection manager:
Responsible for setting up and tearing down the TDM voice channel to
RTP/UDP/IP channel connection.
− SIP protocol stack:
Implements the SIP/UDP/IP protocols for the signalling conversion
and the control of connections.
− Evaluation of the RTCP messages and provisioning of reports to the
softswitch.
− User Port Management:
Responsible for user port administration (setting administrative states)
and operation (monitoring operational states).
• Voice processor
Each TDM voice channel is connected to the voice processor TDM inter-
face during unit configuration. The voice processor provides the following
functions:
− DTMF detection during call setup, notification of detected digits to the
host processor (inband signalling).
The IPSS4 unit supports digit collection according to one or several
configurable digit maps.
− Generation of call progress tones (inband signalling), like dial tone,
ringing tone, busy tone or congestion tone during call setup.
− FSK transmission for analogue display services.
− Setup/teardown of the physical TDM timeslot to packet channel con-
nection.
− Packetize/depacketize TDM voice to/from RTP/UDP/IP packets.
− Voice codec for the conversion of voice codes between TDM and
packet side, i.e. the conversion of the PCM-64 format (G.711, A-law)
to the appropriate format of the packet side.
− Echo cancellation according to G.168: The echo from the telephone
subscriber loop is removed by the digital filter on the transmit path into
the packet network.
− A jitter buffer is provided for each voice channel. The jitter buffer is
used to accommodate the delay variation or misalignment of voice
packets, which would otherwise introduce disturbances on voice sig-
nals.
− T.38 fax relay, i.e. transport of real-time Group 3 fax documents over
IP.
− Silence suppression, i.e. Voice Activity Detection (VAD) and Comfort
Noise Generation (CNG), according to G.711, Appendix II or G.729B.
− Conversion of DTMF tones to RTP packets during a call, i.e. support
of DTMF relay service according to RFC2833 or DTMF transport with
SIP INFO messages.
− Implementation of the Ethernet 1 Gbit/s link interface.

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IPSS4
− RTCP packet generation and reception for round trip delay measure-
ments.
• Subrack internal communication
There is a management link to the COGE5 unit via the backplane which
is based on the KEYMILE proprietary ICN protocol. Software download
and unit control functions are performed over this link.
The internal communication link is implemented in the PBUS of the
MileGate.
• Power
The IPSS4 unit is directly powered from battery (-48 VDC or -60 VDC nom-
inal). Accordingly, a power block is available on the unit for the conver-
sion of the battery voltage to the voltages that are needed on the board.
• GbE star
The IP traffic is transported over two GbE links to the working and the
protecting COGE5 unit.
The links are implemented in the backplane GbE double star of the
MileGate.
2.3.2 Implementation for ISDN-BA access
Please note:
The ISDN-BA subscriber units will be available in a future release.
Figure 4 shows the block diagram of the IPSS4 unit functions for ISDN-BA
access.
Host processor
ISDN signalling NIF & user port management
RTCP handling
TDM / packet
PBUS access
Q.931
connection SIP
Q.921 manager
UDP
HDLC
IP
Media stream
Voice processor
TDM cross connect
Voice activity
Fax detection Tone generator
relay Echo canceller
Voice codec
Subrack internal
Backplane access
RTP RTCP
communication
UDP
IP
SIP
Power RTP stream
GbE star interface

GbE star
Ethernet
Figure 4: IPSS4 functional block diagram for ISDN-BA access

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IPSS4
The functions and components for the ISDN-BA access are the same as for
the PSTN access, except the differences listed below:
• PBUS access
The IPSS4 has a connection capacity of 32 x 2 Mbit/s, providing access
for up to 304 ISDN-BA plus up to 304 PSTN subscribers.
• TDM cross-connect
The TDM cross-connect functionality provides the cross-connections for
the B-channels and the IC-channels (one 128 kbit/s IC-channel carrying
up to 16 D-channels) between ISDN linecards and the IPSS4 unit.
The TDM cross connect separates the voice from the ISDN signalling
data. It directs the TDM voice traffic to the voice processor and the sig-
nalling traffic to the host processor.
• Host processor
ISDN signalling nodal interworking function (NIF):
Responsible for the termination of the Q.921 (data link layer) and Q.931
(network layer) ISDN signalling messages received in the IC-channels
and conversion to SIP commands and vice versa.

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Functions and specifications User Manual
IPSS4
3 Functions and specifications

The IPSS4 unit provides the following functions and conforms to the corre-
sponding standards and recommendations (conformance to applicable parts
of the standards).
3.1 Feature licences
This unit is subject to one or several feature licences. The following licences
are available for this unit.
Table 3: Feature licences relevant for this unit

Licence ID Short Description Description
MFL_T38 MG Lic IPSxx T.38 in VoIP networks MileGate Feature Licence for T.38 (fax) for VoIP gateways -
right to use per network element.
MFL_LCR MG Lic Local Call Routing MileGate Feature Licence for Local Call Routing (also known
as Emergency Stand Alone). Local Call Routing allows calls
between subscribers in the same gateway - right to use per
card.
For more information on features licences please refer to [012] Release Note
“MileGate R4C” and to [915] Technical Bulletin “Feature Licences for
MileGate”.
3.2 Summary of standards
Table 4: Standards
Feature Rating or standard
SIP protocol IETF RFC3261 (06/2002)
- SIP: Session Initiation Protocol
IETF RFC3264 (06/2002)
- An Offer/Answer Model with Session Description Protocol (SDP)
IETF RFC3265 (06/2002)
- Session Initiation Protocol (SIP)-Specific Event Notification
IETF RFC3515 (04/2003)
- The Session Initiation Protocol (SIP) Refer Method
IETF RFC3966 (12/2004)
- The tel URI for Telephone Numbers
IETF RFC4566 (07/2006)
- SDP: Session Description Protocol
SIP authentication IETF RFC2617 (06/1999)
- HTTP Authentication: Basic and Digest Access Authentication
Locating SIP Servers IETF RFC3263 (06/2002)
- Session Initiation Protocol (SIP): Locating SIP Servers

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Table 4: Standards (continued)

Packet transport IEEE 802.1D (1998)
- Media Access Control (MAC) Bridges
IEEE 802.3 (2002)
- CSMA/CD Access Method and Physical Specifications
IETF RFC1213 (05/1991)
- Management Information Base for Network Management of TCP/IP-based inter-
nets: MIB-II
FQDN (fully qualified domain IETF RFC1035 (11/1987)
name) - Domain names - implementation and specification
Ethernet VLAN IEEE 802.1Q (1998)
- Virtual Bridged Local Area Networks
Ethernet priority IEEE 802.1p
- Traffic Class Expediting and Dynamic Multicast Filtering (in 802.1D-1998)
RTP transport IETF RFC2198 (09/1997)
- RTP Payload for Redundant Audio Data
IETF RFC3389 (09/2002)
- Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
IETF RFC3550 (07/2003)
- RTP: A Transport Protocol for Real-Time Applications
IETF RFC3551 (07/2003)
- RTP Profile for Audio and Video Conferences with Minimal Control
RTCP IETF RFC3550 (07/2003)
- RTP: A Transport Protocol for Real-Time Applications
PSTN user support ETSI TS 202 020 (02/2002)
- Harmonized Pan-European/North-American loss and level plan for voice gateways
to IP based networks
ITU-T Q.24 (11/1988)
- Multi frequency push-button signal reception
ISDN-BA user support ITU-T Q.921 (09/1997)
- ISDN user-network interface – Data link layer specification
ITU-T Q.931 (05/19978)
- ISDN user-network interface layer 3 specification for basic call control
IETF RFC4233 (01/2006)
- ISDN Q.921-User Adaptation Layer
ISDN-BA user support 3GPP TS 24.647 V8.4.0 (2010-09)
- Advice Of Charge (AOC) using IP Multimedia (IM) Core Network (CN) subsystem
(Release 8)
ETSI TS 183 036 V3.4.1 (2011-02)
- TISPAN; ISDN/SIP interworking; Protocol specification
ETSI EN 300 182-1 V1.3.6 (1999-09)
- Advice of Charge (AOC) supplementary service; Digital Subscriber Signalling Sys-
tem No. one (DSS1) protocol; Part 1: Protocol specification

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IPSS4
Table 4: Standards (continued)

Codecs and silence suppression ITU-T G.711 (1988)
- Pulse Code Modulation (PCM) of voice frequencies
ITU-T G.711, Appendix II (02/2000)
- A comfort noise payload definition for ITU-T G.711 use in packet-based multimedia
communication systems
ITU-T G.729 (03/96)
- Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-exited linear-
prediction (CS-ACELP)
ITU-T G.729, Annex A (11/96)
- Reduced complexity 8 kbit/s CS-ACELP speech codec
ITU-T G.729, Annex B (11/96)
- A silence compression scheme for G.729 optimized for terminals conforming to
Recommendation V.70
Fax, modem and data transport ITU-T T.4 (07/2003)
- Standardization of Group 3 facsimile terminals for document transmission
ITU-T T.30 (09/2005)
- Procedures for document facsimile transmission in the general switched telephone
network
ITU-T V.17 (02/1991)
- A 2-wire modem for facsimile applications with rates up to 14 400 bit/s
ITU-T T.38 (03/2002)
- Procedures for real-time Group 3 facsimile communication over IP networks
IETF RFC4040 (04/2005)
- RTP Payload Format for a 64 kbit/s Transparent Call
IETF RFC2833 (05/2000)
- RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
Echo canceller ITU-T G.168 (06/2002)
- Digital network echo cancellers

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3.3 Main functions and specifications
Table 5: Main functions and specifications

Feature Rating or standard Release
Maximum number of PSTN user ports 912 r1a
Maximum number of PSTN portgroups 50 r1a
Maximum number of ISDN-BA user ports 304 r1a
Maximum number of ISDN-BA portgroups 20 r1a
Maximum number of simultaneous voice Up to 200 a active voice channels r1a
channels with the G.711 codec
Maximum number of simultaneous voice 80 (+120 G.711 voice channels) r1a
channels with the G.729A codec
Traffic handling capacity 9’750 BHCA r1a
at a CCR > 99.99%
Peak load up to 23 call attempts during 5 s at a CCR > 99.99% r1a
Maximum number of simultaneous T.38 fax 32 r1a
connections
Voice codecs - G.711 codec and support of silence suppression r1a
- G.729A codec and support of silence suppression b
Echo canceller Adaptive echo canceller eliminates echoes from the TDM side r1a
Echo canceller tail length 128 ms r1a
Jitter buffer Adaptive jitter buffer eliminates delay variation from the packet r1a
side
Jitter buffer size up to 200 ms r1a
Fax, modem and data transport - Inband transport of modem- and fax-data with the G.711 r1a
codec.
- Fax relay service according to T.38.
- Transport of 64 kbit/s Clear Mode Data.
- DTMF relay according to RFC2833.
Digit map configuration ITU-T H.248.1 v1 (03/2002) r1a
Gateway control protocol version 1
Trunk hunting Up to 304 trunk hunting groups with DDI (direct dialling in) for r1a
ISDN-BA subscribers.
Support for up to 1000 DDI numbers per trunk hunting group.
ISDN-BA advice of charge (AOC) supple- Charging information at call set-up time (AOC-S) r1a
mentary service Charging information during the call (AOC-D)
Charging information at the end of the call (AOC-E)
Proxy and registrar handling - FQDN (fully qualified domain name) for proxy and registrar r1a
entries with a DNS server
- Support of primary and secondary outbound proxy.
- Support for preferred and alternate DNS server.
- Support for locally provisioned A and SRV entries.
- Support of penalty box for unavailable destinations.
- Support of “locating SIP servers”.
- Local call handling in case of proxy outage.

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Table 5: Main functions and specifications (continued)

Packet transport - Via the GbE star on the backplane of the MileGate subrack r1a
to/from the COGE5 unit
- IP QoS, DSCP for the control and media paths.
- VLAN tagging according to 802.1Q for control and media
paths.
- Priority tagging according to 802.1p for control and media
paths.
RTP transport RTP transport protocol for voice and data r1a
RTP packet size - 5, 10, 20, 30, 40 ms for G.711
- 10, 20, 30, 40 ms for G.729
RTCP RTCP support for maintenance data collection future
release
Profiles - For the PSTN and ISDN-BA user port supplementary ser- r1a
vices and media path configuration.
- For the SIP gateway, Proxy/Registrar and Codec/SDP config-
uration.
- For the enterprise supplementary services for PSTN user
ports.
Alarm reporting ITU-T X.733 (1992) r1a
Information technology – open systems interconnection – sys-
tems management: Alarm reporting function
Performance monitoring - Call statistics r1a
- SIP statistics
- RTP MIB
Event reporting - Alarm events written into the NE and unit alarm logbooks, r1a
- Notifications written into the NE event logbook,
- Protocol errors written into the unit protocol error logbook
1:1 equipment protection r1a
Power supply
- Power supply range VBAT refer to [201] System Description “MileGate R4C”
- Maximum current consumption, IVBAT 0.2 A
VBAT = -48 V
- Maximum total power requirement from 10 W
battery, PTOT
VBAT = nominal voltage
Mechanical parameters
- Construction practice 19 inch
- Height of unit (1 HU = 44.45 mm) 6 HU
- Width of unit (1 TE = 5.08 mm) 4 TE (1 slot)
- Size of the PCB (H x D) 233 mm x 220 mm
- Weight 420 g
- RoHS Directive 2002/95/EC of the European Parliament and of the
Council of 27.1.2003 on the Restriction of the use of certain
hazardous substances in electrical and electronic equipment
- WEEE Directive 2002/96/EC of the European Parliament and of the
Council of 27.1.2003 on waste electrical and electronic equip-
ment

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IPSS4
Table 5: Main functions and specifications (continued)

Reliability
- Calculated MTTF at 35 °C (MIL-HDBK- 84 years
217F)
Emission refer to [201] System Description “MileGate R4C”
Immunity refer to [201] System Description “MileGate R4C”
Safety refer to [201] System Description “MileGate R4C”
Ambient conditions refer to [201] System Description “MileGate R4C”
a. The IPSS4 unit supports up to 80 active voice channels when using the G.729A codec. The G.711 codec is used for
the remainder of the channels up to 200.
b. Please note, that the licence certificate for the use of the G.729AB codec is included in the unit price.
3.4 Call progress tones
The following progress tones can be generated by the IPSS4 unit. Other
tones can be added on request:
• Dial tone
• Ringback tone
• Busy tone
• Congestion tone
• Call waiting tone
• Queued tone (= caller waiting tone)
• Recall dial tone
• Confirmation tone
• Held tone

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IPSS4
4 Installation
This chapter presents the prerequisites for the installation of the IPSS4 unit
in the section 4.1 "Prerequisites" (on page 22).
The section 4.2 "Slots for the IPSS4 unit" (on page 22) shows the available
slots for the unit.
Section 4.3 "Compatibility" (on page 23) gives information about the compat-
ibility with other MileGate units while section 4.4 "Connections and cables"
(on page 23) handles any external connections and cables.
4.1 Prerequisites
Before installing a IPSS4 unit take care to follow the safety advice as listed
in section 1.1 "Precautions and safety" (on page 7).
Valid combinations of hardware (HW) and embedded software (ESW) ver-
sions are given in [012] Release Note “MileGate R4C”.
For the installation of MileGate HW, refer to [301] User Guide “MileGate
2510 Installation” or refer to [310] User Guide “MileGate 2310 Installation”.
Please note:
The hardware of the IPSS4 functional unit is called IPSX4. This name is also
printed on the sticker on the unit front.
4.2 Slots for the IPSS4 unit
The IPSS4 unit uses one slot in the MileGate subrack. Up to two IPSS4 units
can be operated in a MileGate subrack, either as an equipment protection
(EQP) pair or as two independent SIP gateways.
In a MileGate 2510, the IPSS4 unit can be operated in any of the following
slots:
1 … 10, 12 … 21.
In a MileGate 2310, the IPSS4 unit can be operated in any of the following
slots:
7… 10, 12 … 14.
Slot 11 is reserved for the working COGE5 unit.
A protecting IPSS4 unit can be operated in any free slot of the MileGate sub-
rack.

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IPSS4
4.3 Compatibility
4.3.1 Hardware and ESW compatibility
The ESW revision ipss4_r1 is compatible with the hardware unit IPSX4 used
for the IPSS4 functional unit.
4.3.2 MileGate units
The IPSS4 unit is compatible with any other MileGate service unit with an
ESW release of the MileGate system release R4C or later. Please refer to
[012] Release Note “MileGate R4C”
4.3.3 Previous ESW revisions
The IPSS4 unit with ESW release ipss4_r2a is the first release for the
MileGate system release R4C.
4.4 Connections and cables
4.4.1 Front connector of the IPSS4 unit
The IPSS4 unit has no front connectors.
4.4.2 Cable overview
The IPSS4 unit requires no cabling.

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Functional Description User Manual
IPSS4
5 Functional Description
This section gives the detailed functional description of the IPSS4 unit as a
SIP gateway in the MileGate subrack:
• Section 5.1 "Applications" (on page 24) shows some basic applications
using the IPSS4 unit as a SIP gateway.
• Section 5.2 "SIP protocol" (on page 28) presents the functions and fea-
tures of the SIP protocol.
• The basic functions and protocols of the gateway are presented in sec-
tion 5.3 "Basic functions and protocols" (on page 36).
• Section 5.4 "Specific gateway (GW) functions" (on page 41) describes all
the gateway functions as e.g. the media path, the SIP functions or the
supplementary services.
• The different deployment scenarios for proxy servers and registrar serv-
ers can be found in section 5.5 "Proxy server and registrar server func-
tions" (on page 77).
• Section 5.6 "Direct dialling in and trunk hunting" (on page 86) handles the
ISDN-BA specific DDI and trunk hunting features.
• Section 5.7 "Equipment protection (EQP)" (on page 88) handles the con-
figuration and status features of the equipment protection.
• Section 5.8 "Synchronization" (on page 92) handles the synchronization
demand.
• The last section describes the 5.9 "Ethernet switch" (on page 92).
5.1 Applications
5.1.1 MileGate media gateway solutions
The MileGate offers the user a great flexibility to tailor his access gateway or
TDM access network element to the actual need. This makes best use of the
available infrastructure and transport capacity and is most important in case
of migration of existing PSTN and ISDN installations into NGNs.
The main function of the access gateway is implemented in the IPSS4 unit
while the PSTN or ISDN-BA subscriber access is located in the appropriate
linecard units (see section 5.3.1 "SIP with PSTN subscribers" (on page 36)
and section 5.3.2 "SIP with ISDN-BA subscribers" (on page 39)).
In areas with low subscriber density the subscriber linecards can be sub-
tended to locations close to the subscribers, with access to the centralized
media gateway via P12 transport links.
The application examples below show variations on the PSTN and ISDN-BA
side. The media gateway unit is an IPSS4 unit. The NGN applications use
the GbE interface on the COGE5 unit as uplink to the packet switched net-
work.

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IPSS4
The maximum number of subscriber ports the media gateway unit IPSS4
can handle is limited to 912 bearer channels:
• 912 PSTN (POTS) subscribers, or
• 304 ISDN-BA subscribers (2 bearer channels per subscriber) plus 304
PSTN (POTS) subscribers, or
• any mixture of PSTN and ISDN-BA subscribers.
MileGate 2510
Slot Slot Slot Slot
1 11 19 21
S S S S S S S S S S C S S S I S S
U U U U U U U U U U O U U U P U U
P P P P P P P P P P G P P P S P P
M M M M M M M M M M E M M M S M M
5 5 5 5 5 5 5 5 5 5 5 5 5 5 4 5 5
GbE
PSTN PSTN
Port 1 IP Port 912
network
Figure 5: MileGate 2510 equipped for 912 PSTN subscribers with 15

SUPM5 units
MileGate 2310 MileGate 2510

Slot Slot Slot Slot Slot Slot Slot
7 11 14 1 11 19 21
S S S S C S S L L S S S S S S S S S C S S S S I
U U U U O U U O O U U U U U U U U U O U U U U P
I I I I G I I M M I I I I I I I I I G I I I I S
x x x x E x x I I x x x x x x x x x E x x x x S
1 1 1 1 5 1 1 8 8 1 1 1 1 1 1 1 1 1 5 1 1 1 1 4
7 x E1
GbE
IP
ISDN-BA ISDN-BA TDM ISDN-BA network ISDN-BA
Port 1 Port 96 network Port 97 Port 304
Figure 6: MileGate 2310 and MileGate 2510 equipped for 304 ISDN-BA
subscribers with 19 SUIx1 units and one centralized IPSS4 unit

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5.1.2 Gateway capacities
5.1.2.1 PSTN gateway
The number of PSTN service units that can be plugged in a MileGate sub-
rack is limited by the subrack type and if core unit and/or SIP gateway unit
redundancy is used.
Table 6 shows the maximum number of PSTN user ports for some typical
deployment scenarios.
Table 6: PSTN gateway capacities
MileGate COGE5 IPSS4 SIP PSTN PSTN user ports
subrack core unit gateway unit service units with SUPM5
2510 1 1 19 19 x 64 > 912 a
2 2 17 17 x 64 > 912 a
2310 1 1 6 6 x 64 = 384
2 2 4 4 x 64 = 256
a. This scenario reaches the SIP gateway capacity of 912 PSTN user ports
5.1.2.2 ISDN-BA gateway
Please note:
The number of ISDN-BA service units that can be plugged in a MileGate
subrack is limited by the subrack type and if core unit and/or SIP gateway
unit redundancy is used.
Table 7 shows the maximum number of ISDN-BA user ports for some typical
deployment scenarios.
Table 7: ISDN-BA gateway capacities
MileGate COGE5 IPSS4 SIP ISDN-BA ISDN-BA user
subrack core unit gateway unit service units ports with SUIx1
2510 1 1 19 19 x 16 = 304 a
2 2 17 17 x 16 = 272
2310 1 1 6 6 x 16 = 96
2 2 4 4 x 16 = 64
a. This scenario reaches the SIP gateway capacity of 304 ISDN-BA user ports
5.1.2.3 PSTN and ISDN-BA gateway
Please note:
The IPSS4 SIP gateway supports any mixture of PSTN and ISDN-BA user
ports up to a maximum capacity of 912 subscriber channels:
• A PSTN user port has 1 subscriber channel.
• An ISDN-BA user port has 2 subscriber channels.

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The number of PSTN and ISDN-BA service units that can be plugged in a
MileGate subrack is limited by the subrack type and if core unit and/or SIP
gateway unit redundancy is used.
Table 8 shows the maximum number of PSTN and ISDN-BA user ports for
some typical deployment scenarios.
Table 8: PSTN and ISDN-BA gateway capacities
MileGate COGE5 IPSS4 SIP PSTN ISDN-BA PSTN user ISDN-BA user Subscriber chan-
subrack core unit gateway unit service service ports with ports with nels
units units SUPM5 SUIx1
25x0 1 1 15 0 15 x 64 = 960 0 x 16 = 0 960 + 2 x 0 > 912 a
12 5 12 x 64 = 768 5 x 16 = 80 768 + 2 x 80 > 912 b
8 11 8 x 64 = 512 11 x 16 = 176 512 + 2 x 176 = 864
4 15 4 x 64 = 256 15 x 16 = 240 256 + 2 x 240 = 736
2 2 15 0 15 x 64 = 960 0 x 16 = 0 1024 + 2 x 0 > 912 a
12 5 12 x 64 = 768 5 x 16 = 80 768 + 2 x 80 > 912 b
8 9 8 x 64 = 512 9 x 16 = 144 512 + 2 x 144 = 800
4 13 4 x 64 = 256 13 x 16 = 208 256 + 2 x 208 = 672
2310 1 1 4 2 4 x 64 = 256 2 x 16 = 32 256 + 2 x 32 = 320
2 4 2 x 64 = 128 4 x 16 = 64 128 + 2 x 64 = 256
2 2 2 2 2 x 64 = 128 2 x 16 = 32 128 + 2 x 32 = 192
a. This scenario reaches the SIP gateway capacity of 912 PSTN user ports
b. This scenario reaches the SIP gateway capacity of 912 subscriber channels

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5.2 SIP protocol
SIP is a lightweight, transport independent, peer-to-peer protocol. As such it

requires a simple core network where the intelligence is distributed to the
network edge, embedded in endpoints. Endpoints can be SIP terminals built
in either hardware or software, or SIP access gateways (e.g. MileGate with
IPSS4).
To work as a practical service SIP requires proxy and registrar network ele-
ments (servers). Although two SIP endpoints can communicate without any
intervening SIP infrastructure, which is why the protocol is described as
peer-to-peer, this approach is impractical for a public service.
SIP in MileGate relies on the usage of SIP proxy and registrar servers.
5.2.1 SIP connection model
The connection between the TDM and the packet side is managed by the
gateway under the control of the SIP protocol.
In a simple PSTN call, as shown in Figure 7, the voice channel from a
switched circuit network (= SCN bearer channel, e.g. analogue line) is con-
nected to the media stream from a packet network (e.g. RTP stream in an IP
network).
Softswitch
Proxy Registrar
SIP
Media Gateway MG1 Media Gateway

MG2
SIP control
IP
Network
SCN Bearer
Channel  RTP
Stream
NGN voice traffic
SIP telephone
Figure 7: Example of the SIP connection model
Intra gateway calls are handled in the TDM domain, i.e. there is no packet
conversion of the TDM voice signal.
SIP is responsible for the connections between the TDM and packet
resources. SIP controls also the packet side with its transactions and the
exchange of SDP (Session Description Protocol) messages.
The SIP protocol messages are expressed as a series of transactions.
Transactions are transported from gateway to gateway or from gateway to a
SIP telephone.
5.2.2 SIP transactions
SIP is structured as a layered protocol, relying on the UDP/IP transport layer.

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IPSS4
Transaction user
SIP Transaction
Client Server
UDP
IP
Ethernet
Figure 8: SIP layers
Above the transport layer is the transaction layer. The transaction layer has
a client component (referred to as a client transaction) and a server compo-
nent (referred to as a server transaction).
A SIP transaction consists of a single request and any responses to that
request. The client transaction sends the request, and the server transaction
sends the response.
The layer above the transaction layer is called the transaction user (TU). The
IPSS4 gateway as a SIP end system plays the role of a user agent, i.e. is
also a transaction user. When a TU wishes to send a request, it creates a cli-
ent transaction instance and passes it the request along with the destination
IP address, port, and transport to which to send the request.
In the example below, MG1 as the UA client executes the client transaction,
and its Outbound Proxy executes the server transaction. The Outbound
Proxy also executes a client transaction, which sends the request to a server
transaction in the Inbound Proxy. That proxy also executes a client transac-
tion, which in turn sends the request to a server transaction in MG2 as the
UA server.
User Agent (MG1) Outbound Proxy Inbound Proxy User Agent (MG2)
Transaction user Transaction user Transaction user Transaction user

Request Request Request
Client Server Client Server Client Server
transaction Response transaction transaction Response transaction transaction Response transaction
Figure 9: SIP transactions
Each SIP message includes a number of message headers. These headers

contain information for the receiver to know about the call identifier, sender
address, receiver address, request type and other items.
Some headers can be used in both requests and responses. Such headers
contain basic information. For example, the “To:” header field indicates the
recipient of the request, “From:” indicates the originator of the request, and
Call-ID: uniquely identifies a specific invitation to a session.
Other headers are specific for request or response messages.
5.2.2.1 Requests
A request is a SIP message sent from a client to a server, for the purpose of
invoking a particular operation.

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IPSS4
IPSS4 supports the following requests:

• REGISTER
Used by a client to login and register its address with a SIP registrar
server.
• INVITE
Initiates a session. This method includes information about the calling
and called users and the type of media that is to be exchanged.
INVITE is a three way handshake using an ACK request (INVITE –
200 (OK) – ACK), while all other requests are just two way handshakes
without ACK.
• ACK
Sent by the client who sends the INVITE after a final response. ACK is
sent to confirm that the session is established. Media can then be
exchanged.
• CANCEL
Terminates a pending request, such as an outstanding INVITE. After a
session is established, a BYE method needs to be used to terminate the
session.
• BYE
Terminates a session. This method can be sent by either user.
• INFO
Is used for transferring information during a session, such as user activ-
ity.
IPSS4 requires this request for the heart beat (ping) response.
• REFER
Enables the sender of the request to instruct the receiver to contact a
third party using the contact details provided in the request.
IPSS4 requires this request for the call transfer service.
• NOTIFY
The NOTIFY method is used to inform the user that a specific event has
occurred.
IPSS4 requires this request for the call transfer service.
• PRACK
Is an acknowledge of provisional responses used with the reliable deliv-
ery of provisional response messages (option tag “100rel” according to
RFC3262).
• OPTIONS
Queries the capabilities of a server or other devices. It can be used to
check media capabilities before issuing an INVITE.
IPSS4 will reply with a 200 (OK) response without SDP on an OPTION
request.
• UPDATE
Allows a client to update parameters of a session (such as the set of
media streams and their codecs) but has no impact on the state of a dia-
logue. UPDATE can be sent before the initial INVITE has completed.
5.2.2.2 Responses
A response is a SIP message sent from a server to a client, for indicating the
status of a request sent from the client to the server.

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IPSS4
Responses in SIP use a three-digit code followed by a descriptive phrase,

e.g. 301 (moved permanently). SIP distinguishes between provisional and
final responses.
Provisional responses are used by the server to indicate progress, but do not
terminate a SIP transaction. The three-digit code of a provisional response is
1xx.
Final responses terminate a SIP transaction, as opposed to a provisional
response. All 2xx, 3xx, 4xx, 5xx and 6xx responses are final.
On a single SIP request a user agent client expects to receive
• zero or more provisional responses (1xx) and
• one or more final responses (2xx to 6xx).
IPSS4 supports the following responses:
• 1xx: Provisional -- request received, continuing to process the request:
100: Trying
180: Ringing
181: Call is being forwarded
182: Queued
183: Session progress
• 2xx: Success -- the action was successfully received, understood, and
accepted:
200: OK
202: Accepted (required by IPSS4 for call transfer)
• 3xx: Redirection - further action needs to be taken in order to complete
the request
• 4xx: Client Error (request failure) - the request contains bad syntax or
cannot be fulfilled at this server
• 5xx: Server Error - the server failed to fulfil an apparently valid request
• 6xx: Global Failure - the request cannot be fulfilled at any server.
5.2.3 PSTN call establishment and release with SIP
The example scenario, shown in Figure 10 to Figure 12, illustrates how a

PSTN call is established and released with SIP, i.e. how the elements of the
protocol are used for the handling of calls. This example assumes that the
parties involved in the call are connected to two MGs which are connected to
two SIP proxy servers, as shown in Figure 10.
There is another scenario, which is not further discussed here:
• If one party is connected to an MG and the other is connected to a class
5 switch, signalling information has to be exchanged between the softs-
witch and the PSTN via a signalling gateway by means of SIP.

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IPSS4
Softswitch 1 Softswitch 2
„provider1.com“ „provider2.com“
IP: 111.111.111.111 IP: 222.222.222.222
Proxy Registrar Proxy Registrar
SIP SIP
Media Gateway MG1 Media Gateway MG2

IP: 111.111.111.100 IP IP: 222.222.222.200
Network
1111 2222
NGN voice traffic
Figure 10: NGN network set up for the illustration of call establishment and
call release
In the example of Figure 10, it is assumed, that both MGs have idle ana-
logue line (physical) terminations programmed to look for call initiation
events (i.e. off hook). The subscriber numbers of the analogue line are 1111
in MG1 and 2222 in MG2. The allocation of these subscriber numbers to the
appropriate device URI has to be registered in the corresponding softs-
witches (Registrar). The registering is initiated by the gateway.
Figure 11 shows an example for the transfer of information between the ana-
logue phone, the MGs and the proxy servers during the establishment of a
call. The following steps are executed (step numbers in Table 9 correspond
to the numbers related to requests and responses in Figure 11).
Table 9: Call establishment with SIP
Step Description
1 MG1 detects an off-hook event from subscriber 1111.
2 MG1 plays the dial tone towards subscriber 1111. It looks for digits accord-
ing to a configured dial plan and looks for an on-hook event.
3 Digits are accumulated by MG1 as they are dialled by subscriber 1111. The
dial tone is stopped upon detection of the first digit. When an appropriate
match is made of collected digits against one of the valid dialling plans, an
INVITE request is sent to the SIP proxy server that serves the subscriber
1111 domain (provider1.com) using the SIP URI of subscriber 2222.
4 The SIP proxy server receives the INVITE request and sends a 100 (Try-
ing) response back to MG1. The 100 (Trying) response indicates that the
INVITE has been received and that the proxy is routing the INVITE to the
destination.
5 The provider1.com proxy server locates the proxy server at provider2.com,
and forwards, or proxies, the INVITE request there.
Before forwarding the request, the provider1.com proxy server adds an
additional Via header field value that contains its own address (the INVITE
already contains MG1’s address in the first Via).
6 The provider2.com proxy server receives the INVITE request and responds
with a 100 (Trying) response back to the provider1.com proxy server to
indicate that it has received the INVITE and is processing the request.
7 The provider2.com proxy server knows the location (IP address) of sub-
scriber 2222 from the registrar server, adds a further Via header line with its
own address and proxies (forwards) the INVITE request to MG2.
8, 9 MG2 receives the INVITE and alerts subscriber 2222 to the incoming call
from subscriber 1111 with a ringing signal, so that subscriber 2222 can
decide whether to answer the call or not.

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IPSS4
Table 9: Call establishment with SIP (continued)

Step Description
MG2 indicates this in a 180 (Ringing) response, which is routed back
through the two proxies in the reverse direction. Each proxy uses the Via
header field to determine where to send the response and removes its own
address from the top.
10 When MG1 receives the 180 (Ringing) response, it applies a ringback tone
to subscriber 1111.
11 Subscriber 2222 decides to answer the call. When he picks up the handset,
MG2 stops the ringing signal towards subscriber 2222 and sends a 200
(OK) response towards the proxy to indicate that the call has been
answered.
The 200 (OK) response contains a contact header field with an URI at
which MG2 can be reached directly.
MG2 makes now the connection between the circuit switched bearer chan-
nel of subscriber 2222 and the packet resources.
12 The 200 (OK) is routed back through the two proxies and is received by
MG1, which then stops the ringback tone.
MG1 makes now the connection between the circuit switched bearer chan-
nel of subscriber 1111 and the packet resources.
13 Finally, MG1 sends an acknowledgement message (ACK) to MG2 to con-
firm the reception of the final response (200 (OK)).
In this example, the ACK is sent directly from MG1 to MG2, bypassing the
two proxies because the gateways have learned each other’s address from
the contact header fields through the INVITE/200 (OK) exchange.
Subscriber 1111 and subscriber 2222 are now connected via MG1 and
MG2 with a media session.

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IPSS4
Media Gateway MG1 „provider1.com“ „provider2.com“ Media Gateway MG2
1111 2222
1 Off hook 1
2 Dial tone 2
3 Digits 3
INVITE
4 100 (Trying) 4
5 INVITE 5
6 100 (Trying) 6
7 INVITE 7
8 Ringing 8
9 180 (Ringing) 9
180 (Ringing)
180 (Ringing)
10 Ring back tone 10
11 Off hook 11
OK
12 OK 12
OK
13 ACK 13
Figure 11: Call establishment with SIP
Figure 12 shows the transfer of information between the analogue phone,

the MGs and the proxy servers during the release of a call. The following
steps are executed (step numbers in Table 10 correspond to the numbers
related to events and commands in Figure 12).
Table 10: Call release with SIP
Step Description
1 Subscriber 2222 hangs up. Accordingly, an on hook event is detected by
MG2.

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IPSS4
Table 10: Call release with SIP (continued)

Step Description
2 MG2 generates a BYE message. This BYE is routed directly to MG1, again
bypassing the proxies.
MG2 releases the connection between the circuit switched bearer channel
of subscriber 2222 and the packet resources and is ready now for the next
off-hook event.
3 MG1 confirms the receipt of the BYE with a 200 (OK) response, which ter-
minates the session and the BYE transaction.
MG1 releases the connection between the circuit switched bearer channel
of subscriber 1111 and the packet resources.
(No ACK is sent by MG2. ACK is sent only in response to a 200 (OK)
response from an INVITE request.)
4 Subscriber 1111 hangs up. Accordingly, an on hook event is detected by
MG1.
MG1 is ready now for the next off-hook event.
Media Gateway MG1 „provider1.com“ „provider2.com“ Media Gateway MG2
1111 2222
1 On hook 1
2 BYE 2
3 200 (OK) 3
4 On hook 100 (Trying) 4
Figure 12: Call release with SIP

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IPSS4
5.3 Basic functions and protocols
5.3.1 SIP with PSTN subscribers
SUPMx IPSS4 COGE5

PSTN signalling
(out of band ) PSTN signalling PSTN signalling interworking VLAN bridging
(out of band ) function & Traffic prioritisation SIP
to V5CAS connection control
Softswitch
Ethernet star
DTMF Call control
Con-
PBUS
(inband tones
nection
signal - (inband
control
ling) signalling)
RTP
Voice channel PCM64 codec VLAN bridging packet
TDM stream
(with inband (G.711 A-law) Traffic prioritisation MG
PSTN signalling ) Packet
Voice traffic protocol stack

RTP RTP
TDM based
voice channel UDP UDP
TDM based voice channel IP 802.1Q IP

analogue line PBUS PBUS Ethernet Ethernet Ethernet Ethernet
Signalling /control protocol stack

NIF
SIP SIP
PSTN
UDP UDP
signalling
PSTN signalling IP 802.1Q IP
analogue line PBUS PBUS Ethernet Ethernet Ethernet Ethernet
signalling data
connection control
voice traffic
Figure 13: Basic function and protocol view for signalling and voice traffic in
the case of PSTN access
Figure 13 shows a schematic view of the basic PSTN functions with the
appropriate protocols of the access gateway, realised with the IPSS4 unit, a
PSTN linecard and the control unit COGE5.
5.3.1.1 Voice traffic
On the PSTN linecard, the codec converts the analogue voice traffic into a
PCM-64 signal and vice versa, according to ITU-T G.711 (A-law). Voice traf-
fic stands for voice or voice band data from a modem or a fax.
The PCM-64 signal is cross connected between the PSTN linecard and the
IPSS4 unit. The IPSS4 unit implements the conversion between PCM-64
signals and packets. Note, that the IPSS4 unit supports two different codecs
for the transport of voice across the packet network:
• G.711 codec with the PCM A-law companding and support of silence
suppression (voice activity detection and comfort noise generation
according to G.711, Appendix II)

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IPSS4
• G.729A codec and support of silence suppression (voice activity detec-

tion and comfort noise generation according to G.729B).
RTP (Real-Time Transport Protocol, defined in RFC3550) is used for the
transport of voice packets across an IP network. Packet sequence numbers
and timestamps are carried in the RTP header. Packet sequence numbering
allows the user to reorder packets at the receiver side and to detect the loss
of packets. Timestamps allow the detection of jitter. These performance data
are sent to the softswitch at the end of the call and are used for fault and
performance maintenance. For a description of the required cross connec-
tions between the PSTN linecard and the IPSS4 please refer to [314] User
Guide “TDM Services and Cross Connections”.
5.3.1.2 Control traffic
SIP is an application layer control protocol handling the setup, modification

and tear down of multimedia sessions, defined in RFC3261 and described in
section 5.2 "SIP protocol" (on page 28).
In the SIP environment IPSS4 plays the role of a user agent (UA) for every
connected channel while the SIP proxy server and SIP registrar server are
summarized in the term softswitch.
The connection control, call control and resource control functionality is
imbedded in the SIP endpoint, i.e. in the IPSS4. Accordingly, the IPSS4 unit
has a control part that handles signalling and the commands for setting up
and tearing down the TDM voice channel to virtual packet channel connec-
tions.
Please note:
In the application with IPSS4 the SIP sessions are all voice telephony ses-
sions.
An important aspect is the signalling interworking function. PSTN signalling
is converted to SIP requests and vice versa. We have to distinguish between
out of band- and inband signalling.
With out of band signalling (e.g. on-hook and off-hook or pulse dialling in
upstream direction, ringing in downstream direction etc.), the PSTN linecard
converts PSTN signalling information to CAS (Channel Associated Signal-
ling) code and vice versa. CAS in the MileGate is the V5CAS:
V5CAS is a proprietary CAS signalling code which is an extension of MCAS
(Mercury CAS) and carries out of band PSTN signalling. Note that the name
“V5CAS” exists from historical reasons and has no relation to the V5 protocol
suite.
The signalling channel is cross connected between the PSTN linecard and
the IPSS4 unit. On IPSS4, the conversion CAS ↔ SIP is implemented. In
upstream direction (towards the softswitch), the IPSS4 receives out of band
signalling via CAS bits and notifies these events to the softswitch with the
SIP protocol. In downstream direction (towards the analogue line), the IPSS4
receives SIP requests and responses and converts them into CAS coded
messages.
With inband signalling (DTMF dialling in upstream direction, insertion of call
progress tones like dial tone or busy tone in downstream direction), PSTN
signalling information is transported in the voice channel (also called bearer
channel) and terminated in the IPSS4 unit. This means, that DTMF tones are
interpreted in the IPSS4 and converted into SIP requests. On the other hand,

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a SIP request can instruct the IPSS4 unit to transmit e.g. a dial tone towards
the analogue line.
The mapping of SIP requests to PSTN signalling is specified by national pro-
tocols. In MileGate, this mapping is defined with a Custom Parameter Set
(CPS).
Please note:
For a description of the CPS please refer to section 6.1 "Customer profiles"
(on page 93).
The IPSS4 unit supports the following PSTN signalling messages (events
and signals):
• On hook detection
• Off hook detection
• Hook flash detection
• DTMF dialling
• Pulse dialling
• Pulsed no battery
• Cadence ring generation
• Ring splash (initial ring)
• Steady signal normal polarity
• Steady signal reversed polarity
• Call progress tones (e.g. dial tone, busy tone etc.)
• Analogue display using FSK signalling (e.g. for calling line identification
presentation, CLIP)
The TE Alerting Signal (TAS) to be used is pre-provisioned per gateway:
− On-hook CLIP:
TAS is hard coded as cadence ring with the same pattern as alert/ri.
Data is sent in the first ring pause.
− Off-hook CLIP:
TAS is hard coded as Dual Tone Alerting Signal (DT-AS). After the
subsequent data transmission the gateway sends a cg/cw signal to
indicate the completion of the procedure.
• Analogue display using DTMF signalling (e.g. for calling line identification
presentation, CLIP), alternatively used to FSK signalling.
The PSTN signalling messages “Pulsed reduced battery”, “Reduced battery”,
“No battery” and “B-wire connected to earth” could be added on IPSS4, if
required.
Please note:
The control of linecard maintenance functions (e.g. locking of a PSTN user
port and line testing) by the softswitch or a remote user agent is not sup-
ported. The maintenance functions on PSTN linecards are managed by the
ECST.
Both, signalling and voice traffic are transported with UDP/IP over the Ether-
net traffic port of the core unit COGE5 towards the IP network.

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5.3.2 SIP with ISDN-BA subscribers
Please note:
SUIx1 IPSS4 COGE5

ISDN signalling interworking VLAN bridging
Mux / function & Traffic prioritisation SIP
Softswitch
Demux connection control
Ethernet star
S/T-IF D DTMF Call control
PBUS
Con-
(inband tones nection
signal - (inband control
ling) signalling )
RTP
TDM VLAN bridging packet
U-IF 2B+D 2B stream
Termin. Demux Traffic prioritisation MG
U-IF / Mux Packet
Basic Rate Access

Voice traffic protocol stack
TDM based RTP RTP
TDM based
voice voice channel UDP UDP
channel
B-channel B-channel B-channel IP 802.1Q IP
S/T-IF S/T-IF U-IF U-IF PBUS PBUS Ethernet Ethernet Ethernet Ethernet
Signalling/control
protocol stack
Q.931 NIF
Q.931 SIP SIP

Q.921 /
LAPD Mux / Demux Q.921 UDP UDP
D-channel D-channel IC-channel IC-channel IP 802.1Q IP
S/T-IF S/T-IF U-IF U-IF PBUS PBUS Ethernet Ethernet Ethernet Ethernet
signalling data
connection control
voice traffic
Figure 14: Basic function and protocol view for signalling and voice traffic in
the case of ISDN-BA access
Figure 14 shows a schematic view of the basic ISDN-BA functions with the
appropriate protocols of the access gateway, realised with the IPSS4 unit, an
ISDN-BA linecard and the control unit COGE5.
The ISDN basic rate interface is the standard interface to connect ISDN sub-
scribers. Each ISDN circuit includes three channels:
− 2 B- or bearer channels for data or voice (each 64 kbit/s).
− 1 D- or data channel for signalling (16 kbit/s).
The Terminal Equipments (TE) of the subscriber are connected to the NT-1
(Network Termination 1) via S-Interface (or S/T-Interface in the case of the
usage of an NT-2, like e.g. a PABX). It is possible to have up to 8 TEs per
subscriber. There are two types of TEs: ISDN compatible TEs, which are
directly connected to the S-Interface, and traditional analogue TEs, which
are connected to the S-Interface via a Terminal Adaptor (TA). The U-Inter-
face defines the physical connection from the NT-1 to the ISDN line card via
the existing telephone wire pair.

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IPSS4
5.3.2.1 Voice traffic
On the subscriber side, the codec converts an analogue voice signal into a
PCM-64 signal and vice versa, according to G.711 (A-law). A voice signal
includes voice or voice band data from a modem or a fax. This codec is
implemented either in the ISDN compatible TEs or in the Terminal Adaptors,
which are used for the connection of analogue phones to the S-Interface.
The PCM-64 signal is transported to the ISDN-BA linecard across the U-
interface. In the MileGate, it is cross connected between the ISDN-BA line-
card and the IPSS4 unit. On the IPSS4 unit, the conversion between PCM-
64 signals and packets is implemented. Note, that the IPSS4 unit supports
two different codecs for the transport of voice across the packet network:
• G.711 codec with the PCM A-law companding and support of silence
suppression (voice activity detection and comfort noise generation
according to G.711, Appendix II).
• G.729A codec and support of silence suppression (voice activity detec-
tion and comfort noise generation according to G.729B).
RTP (Real-Time Transport Protocol, defined in RFC3550) is used for the
transport of voice packets across an IP network. This conversion is identical
to the appropriate function with PSTN access.
5.3.2.2 Signalling / Control traffic
The D-channel transports signalling data and uses both the link layer
(= layer 2) and the network layer (= layer 3). The link layer and the network
layer protocols are specified in ITU-T Q.921 and ITU-T Q.931 respectively.
The link layer protocol is also called “Link Access Protocol on the D channel”
(LAPD).
Note, that there is one D-channel per NT-1. However, several link layer con-
nections may exist per D-channel, in order to provide one link layer connec-
tion per TE attached to the S-Interface.
The D-channels are cross connected between the ISDN-BA linecard and the
IPSS4 unit via the PBUS. For this purpose, all D-channels of one ISDN-BA
linecard unit are statistically multiplexed into one 128 kbit/s Internal Commu-
nication (IC) channel.
As shown in Figure 14, the Q.921 layer and the Q.931 signalling messages
are terminated on the IPSS4 unit. The signalling messages are converted to
SIP requests and forwarded to the softswitch.
Both, signalling and voice traffic are transported with UDP/IP over the Ether-
net traffic port of the core unit COGE5 towards the IP network.

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5.4 Specific gateway (GW) functions
5.4.1 Media path
5.4.1.1 Voice
The figure below shows the functional blocks that are involved in the voice
conversion and voice handling in the gateway:
Fax/modem
call
detection
DTMF Voice Silence

detection activity insertion
detection descriptor
(VAD ) (SID)
G.711 to Gain Echo Non linear Comfort Voice TDM to

linear setting canceller processor noise coding packet
conversion (EC) (NLP) generation conversion
(CNG)
TDM side Packet side
Linear to Gain Comfort Jitter

G.711 setting noise buffer
conversion generation
Call Packet loss

progress conceal -
tones ment
insertion (PLC)
Comfort SID
noise detection
generation
Voice Packet
decoding to TDM
conversion
Figure 15: Functional blocks in the gateway
TDM to packet direction:

A user signal entering the gateway at the TDM side is first converted from
the G.711 coded signal to a 16 bit linear coded signal, to be able to do all the
required operations on this signal.
In a second step, the signal level can be adjusted in the gain block. Note that
in IPSS4 the gain is fixed to 1, i.e. the level is not changed. A change of the
signal level can be done on the PSTN linecard unit.
The gain adjusted signal is fed to the DTMF detection block (see section
5.4.1.6 "DTMF detection" (on page 46)) and to the fax/modem call detection
block (see section 5.4.1.4 "Fax and modem call detection" (on page 43)).

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IPSS4
The echo canceller function comprises the blocks “Echo canceller”, “Non lin-
ear processor” and “Comfort noise insertion” (see section 5.4.1.8 "Echo can-
celler" (on page 47)).
The silence suppression function comprises the blocks “Voice activity detec-
tion” and “Silence insertion descriptor” (see section 5.4.1.9 "Silence suppres-
sion" (on page 47)).
In the voice coding block, the linear coded signal is converted to the G.711
A-law code or the G.729A code.
Finally the coded user signal is mapped to RTP/UDP/IP packets and for-
warded to the Ethernet interface.
Please note:
In the ISDN-BA clearmode application the user signal remains completely
unchanged.
Packet to TDM direction:
Packets entering the gateway are first stored in the jitter buffer. Missing
packets are detected and the (missing) voice signal is replaced by comfort
noise in the “packet loss concealment” block. Misaligned packets (up to four
positions) are detected and reordered. See section 5.4.1.10 "Jitter buffer"
(on page 49) for the detailed description of the jitter buffer.
Packets containing the silence insertion descriptor are detected and a com-
fort noise signal with the corresponding level is generated.
The G.711 A-law or G.729A coded signal is extracted from the packets con-
taining voice and converted to the 16 bit linear format.
The voice signal can be replaced by call progress tones (see section
5.4.1.11 "Call progress tone generation" (on page 49)).
The signal level can be adjusted in the gain block. Note that in IPSS4 the
gain is fixed to 1, i.e. the level is not changed. A change of the signal level
can be done on the PSTN linecard unit.
Finally the 16 bit linear coded signal is converted to the G.711 coded signal
and forwarded to the TDM interface.
Please note:
In the ISDN-BA clearmode application the user signal remains completely
unchanged.
5.4.1.2 Voice codecs
IPSS4 supports the following voice codecs:

• G.711 with A-law companding and with optional silence suppression
according to G.711, Appendix II. IPSS4 can be configured to use or not to
use silence suppression with the G.711 codec.
• G.729A with silence suppression according to G.729B. IPSS4 can be
configured to use or not to use silence suppression with the G.729A
codec.
Please note:
The licence certificate for the use of the G.729A/B codec is included in the
unit price.

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IPSS4
5.4.1.3 64 kbit/s clearmode data
In the clearmode application, the raw 64 kbit/s data stream is transported,

i.e. all the voice functions like code conversion, echo canceller and voice
activity detection are switched off.
Typical applications are “Internet access over ISDN” or “ISDN fax transmis-
sion”. Clearmode is a pure ISDN function. The IPSS4 unit detects the clear-
mode call in the setup message from the local ISDN-BA terminal and
changes the configuration of the voice path accordingly.
5.4.1.4 Fax and modem call detection
A fax or modem call is detected by the IPSS4 on the callee side when it
receives a
• CED (2100 Hz) followed by V.21 flags (G3 fax).
• ANSam tone (2100 Hz with phase reversals) (super G3 fax or high speed
modem).
When a fax or modem call is detected
• the non linear processor (NLP) of the echo canceller is switched off (2100
Hz tone).
• the echo canceller and the non linear processor are switched off (2100
Hz tone with phase reversals).
• the packet loss concealment (PLC) is enabled.
• silence suppression (SS) with voice activity detection (VAD) and comfort
noise generation (CNG) is disabled.
• the jitter buffer is set to a fixed depth.
• the gain is set to 1.
5.4.1.5 Fax and modem data transport
The IPSS4 unit supports voice band data (VBD) mode for the inband trans-
port of fax and modem data over G.711.
In the case of fax service, T.38 relay can be applied instead of the inband
transport (“Fax over IP“).
VBD fax transport:
In the G3 fax TE, the image data is coded according to ITU-T T.4. The T.4
coded control and image data is transmitted towards the gateway as a VBD
signal. The VBD signal is handled by the gateway as a voice signal, i.e. it is
encapsulated in packets according to the RTP protocol using the G.711
codec.

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IPSS4
G3 FAX TE Access Gateway IP Access Gateway G3 FAX TE

(coding acc. to T.4) (without T.38 relay) Network (without T.38 relay) (coding acc. to T.4)
T.4
T.30
FAX-VBD signal FAX-VBD signal over G.711 FAX-VBD signal
Figure 16: Basic view for fax transmission without T.38 fax relay
The opposite gateway decapsulates the VBD signal from the RTP packets
and forwards it to the remote fax terminal.
IFP internet facsimile protocol (T.38):
In the G3 fax TE, the image data is coded according to ITU-T T.4. The T.4
coded control and image data is transmitted towards the gateway as a VBD
signal. In the gateway, the fax-VBD signal is demodulated and converted
into T.38 format (and vice versa), as shown in Figure 17. The image data are
transferred into IFP packets (Internet Facsimile Protocol, according to T.38)
and transmitted via UDP / UDPTL transport protocols.
G3 FAX TE Access Gateway IP Access Gateway G3 FAX TE

(coding acc. to T.4) (with T.38 relay) Network (with T.38 relay) (coding acc. to T.4)
T.4
T.30
FAX-VBD signal T.38 (over IP) FAX-VBD signal
Figure 17: Basic view for fax transmission with T.38 fax relay
Figure 18 shows the architectural framework of the ITU-T T.38 recommenda-

tion. A traditional Group 3 fax terminal equipment is connected to a gateway,
transmitting a fax through an IP network to a receiving gateway.
• The terminals establish the T.30 session by negotiating the capabilities of
the fax functions i.e. the parameters of the G3 fax TEs.
• After the successful establishment, T.4 coded fax data can be transmitted
from the sending terminal to the receiving terminal.
• In the access gateway, the fax-VBD signal is demodulated and converted
into T.38 format (and vice versa), as shown in Figure 19.

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IPSS4
G3 FAX TE
Coder
Scanner (T.4)
Access Trunking G3 FAX

FAX Gateway IP Gateway PSTN TE
control Modem
(T.30) Network Network
Decoder
Printer (T.4)
Figure 18: Block diagram, showing the T.38 fax relay function in the case of
PSTN access
Ethernet star
SUPMx IPSS4 COGE5
PBUS
IFP
VLAN bridging packet
PCM64 codec (G.711 Modem IFP packet (T.38) stream
Traffic prioritisation
A-law) de/encapsulation MG
T.38 fax relay protocol stack

T.4
T.30
T.38 T.38
Modem Modem
IFP IFP
V.17 V.17
UDP UDP
UDPTL UDPTL
TDM based voice channel IP 802.1Q IP
analogue line analogue line PBUS PBUS Ethernet Ethernet Ethernet Ethernet
Figure 19: Protocol view for T.38 fax relay function in the case of PSTN
access (assuming that V.17 is used as fax transmission proto-
col)
If T.38 fax relay is enabled (by means of configuration) the T.4 coded fax
data is converted to T.38 format on the IPSS4 unit. The image data are
transferred into IFP (internet facsimile protocol) packets and transmitted via
UDP / UDPTL transport protocols. It is possible to have up to 32 simultane-
ous T.38 fax connections on an IPSS4 unit.
A call is set up with audio capability only. When the terminating side detects
a fax tone then it initiates a re-invite to switch to T.38 fax by muting audio
and offering image capability.
With T.38, it is possible to have error recovery by means of transmitting
redundant fax packets. The T.38 UDP redundancy level, i.e. the number of
redundant packets to be sent is 3. This requires the use of the UDPTL proto-
col (UDP transport layer).
Note, that the IPSS4 can detect the type of traffic (i.e. voice and different cat-
egories of fax and modem) by the evaluation of the preamble which is trans-
mitted in the case of fax and modem traffic.
When T.38 is enabled, all G3 fax formats apart from V.34 are converted to
T.38, if the partner GW supports the T.38 relay function. V.34 fax is trans-
ported as VBD.

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IPSS4
If T.38 fax relay is disabled (by means of configuration) the IPSS4 unit will
reject the image offering in a received SDP message and offer audio capa-
bility only. The fax will therefore be transported as voice band data (VBD).
Before switching from G.729 or G.711 to VBD mode IPSS4 can initiate a re-
INVITE offering G.711 including the attribute ‘a=gpmd:8 vbd=yes’. This is
controlled by the CPS parameter SIP1.21.
When the fax transmission is terminated the IPSS4 can switch back to audio.
The IPSS4 initiates a re-invite to switch to audio. This is controlled by the
CPS parameter VP1.37.
5.4.1.6 DTMF detection
DTMF tones from a PSTN phone set are detected in accordance with the
ETSI standard ES 201 235-3, i.e. with the corresponding
• frequency tolerance
• level tolerance, including twist
• tone duration tolerance
• pause duration tolerance
Detected DTMF tones are embedded in the SIP request to the softswitch.
The IPSS4 unit supports several configurable digit maps.
5.4.1.7 DTMF tone transport
DTMF tones can be transported to the remote gateway using different meth-
ods:
• inband, i.e. as voice signals, or
• using the DTMF relay service according to RFC2833, or
• with SIP INFO messages.
The transport mode and the transport method are configurable within the
Codec/SDP profile or the unit configuration AP: /unit-x, Configuration: Gate-
way - Media - Codec/SDP.
The DTMF relay service is supported for both codecs (G.711 and G.729A).
When using DTMF relay service according to RFC2833, the inband transport
of the tones is squelched.
Transmission of RFC2833 packets operates on a per-call basis, after the
capabilities of the peer GW have been identified through the receipt of
appropriate SDPs.
From the received RFC2833 packets the DTMF tones are regenerated with
their frequency, level and duration.
DTMF tone transport is a type of telephone-event transport. The RTP pay-
load format for the telephone-event transport uses a dynamic payload type in
the range 96 to 127. The payload type is configurable within the Codec/SDP
profile or the unit configuration AP: /unit-x, Configuration: Gateway - Media -
Codec/SDP; the default value is 96.
The SIP INFO relay service reports the DTMF tones to the remote end with a
fixed length of 250 ms. The tones are transported inband in parallel.

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5.4.1.8 Echo canceller
An echo canceller is a voice operated device used for reducing the echo pre-
sent on the transmit path. Echo cancellation is required, because of common
echo problems in voice over packet networks due to the usual round trip
delay of greater than 50 ms, which is higher than what PSTN specification
dictates. The delays in the packetization, voice compression, buffering,
switches in the network, etc. all contribute to this delay.
The echo canceller subtracts an estimate of the echo produced by the hybrid
or by the acoustic coupling in the telephone set from the echo signal. This
digital filtering process is adaptive, i.e. it takes several seconds after a call is
setup to remove the echo to a sufficient level.
Transmit path Non linear Comfort
Network Digital processor noise
functions subtractor (NLP) generation
(CNG)
Hybrid Echo
estimator
Network Receive path

functions
Cancelled end Non cancelled end

Figure 20: Functional blocks of the echo canceller
Due to the imperfect cancellation there remains a small amount of echo that
can not be removed. This residual echo is cancelled in the non linear proces-
sor (NLP) by removing the transmit signal completely when the detected
level in the transmit signal is very small.
To prevent the annoyance of speech pauses without background noise, the
comfort noise generation block inserts a pseudo-random noise signal which
level is adapted to the incoming background noise.
There is no parameter for the configuration of the echo canceller function.
The tail length, depending on the maximum delay on the subscriber line, is
adapted automatically.
Note that the properties of the echo canceller are defined by the IPSS4 unit
and cannot be configured by the softswitch or the remote user agent.
5.4.1.9 Silence suppression
The silence suppression function comprises the blocks

• “Voice activity detection”,
• “Silence insertion descriptor (SID) insertion”,
• “Silence insertion descriptor (SID) detection” and
• “Comfort noise generation”.

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Fax/modem
call
detection
DTMF Voice Silence

detection activity insertion
detection descriptor
(VAD ) (SID)
G.711 to Gain Echo Non linear Comfort Voice TDM to

linear setting canceller processor noise coding packet
conversion (EC) (NLP) generation conversion
(CNG)
TDM side Packet side
Linear to Gain Comfort Jitter

G.711 setting noise buffer
conversion generation
Call Packet loss

progress conceal -
tones ment
insertion (PLC)
Comfort SID
noise detection
generation
Voice Packet
decoding to TDM
conversion
Figure 21: Functional blocks for the silence suppression
In a typical voice conversation, one person speaks while the other listens.
There is no need to transmit voice packets when a person is not speaking.
This feature is also referred to as “silence suppression”. Silence suppression
can reduce the average bandwidth required to support a voice conversation
by almost 40%.
Voice activity detection (VAD) monitors the voice data at the encoder. When
no voice activity is detected for a pre-determined period of time, the VAD
declares the voice channel to be inactive (or silent). No voice packets are
sent any more.
Even though no packets are transmitted during silence periods, listeners
connected to the decoder should hear or experience a background noise
similar to what they would hear in the absence of silence suppression
schemes. Comfort noise generation is the mechanism to create this back-
ground noise in a synthetic way. The encoder periodically sends information
about the background noise to the receiver using silence insertion descriptor
(SID) frames.
The silence suppression function with the G.711 codec and with the G.729A
codec can be enabled or disabled per gateway configuration or with the
Codec/SDP profile.

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5.4.1.10 Jitter buffer
Received RTP packets are supervised for their sequence number. Mis-
aligned packets, up to an offset of four packets, are reordered and the voice
part of the received packets is written then to the jitter buffer.
A jitter buffer is provided for each voice channel with a depth of up to
200 ms. The jitter buffer is used to accommodate the delay variation of voice
packets, which would otherwise introduce clicks or other undesirable effects
on voice signals, as well as loss of transmitted data between network
devices. The jitter buffer is essentially a small delay buffer located at the
receiving end of the IP voice connection, which intentionally delays the arriv-
ing packets. The voice packets are collected, stored, and sent to the voice
processor in evenly spaced intervals.
The jitter buffer is operated in an adaptive mode. With the adaptive jitter
buffer, the threshold value is adapted according to the current traffic situa-
tion.
For the jitter buffer, the configuration of the following parameters is required:
• Nominal threshold:
This parameter defines the buffer fill level, where the stored voice pack-
ets are read out after the set up of a call.
The nominal threshold should be set as low as possible, as it also directly
results in delay.
• Fax+clearmode threshold:
This parameter defines the buffer fill level, where the stored fax or clear-
mode data packets are read out after the set up of a call.
Note that during the transport of data the jitter buffer will automatically be
set to “fixed”. Due to the fact that on one hand the transport of data is not
delay sensitive and on the other hand, loss of packets should be avoided,
the threshold for data should be significantly higher compared to the
threshold for voice.
• Low Threshold:
This parameter defines the minimum jitter buffer threshold to be used in a
low packet jitter transport environment.
• High Threshold:
This parameter defines the maximum jitter buffer threshold to be used in
a high packet jitter transport environment.
• Response Time:
This parameter defines the dynamic behaviour of the threshold adapta-
tion. Each time the “response time” timer elapses the new jitter buffer
threshold is calculated from the measured packet jitter values.
Note that the properties of the jitter buffer are defined by the IPSS4 unit and
cannot be configured by the softswitch or the remote user agent.
5.4.1.11 Call progress tone generation
Controlled by the received SIP requests, IPSS4 plays the call progress tones
towards the TDM interface.
IPSS4 supports the call progress tones according to the
• H.248.1 call progress tones generator package (cg),
• Q.1950 basic call progress tones generator with directionality package
(bcg), and

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IPSS4
• Q.1950 expanded call progress tones generator package (xcg).

The tones according to the national variations are defined with the custom
parameter set (CPS). It is e.g. also possible to play different call progress
tones for different 4xx (client error) SIP responses.
5.4.1.12 Bandwidth requirements
The required bandwidth per voice call can be calculated taking into account
the voice duration per packet, the codec in use and the overhead in the
Ethernet packets.
The packet overhead is calculated as follows:
Table 11: Packet overhead
Packet Overhead bytes per
packet with VLAN
Ethernet interframe gap 12
Preamble 8
Ethernet 18
VLAN 4
IP 20
UDP 8
RTP 12
Total 82
IPSS4 supports the two codecs G.711 and G.729A with the following pack-
etization intervals:
• When IPSS4 is the offerer then it will select the IETF recommended
packetization time which is 20 ms for G.711 and G.729.
• If the remote side is the offerer then IPSS4 will apply the proposed pack-
etization time:
− For G.711, IPSS4 supports:
5 ms, 10 ms, 20 ms, 30 ms, 40 ms.
If a value outside this range is offered then IPSS4 will apply 20 ms.
− For G.729, IPSS4 supports:
10 ms, 20 ms, 30 ms, 40 ms.
If a value outside this range is offered then IPSS4 will apply 20 ms.
The example calculations below are made under the assumption of a packet
overhead of 82 octets.
Table 12: Packet bandwidth
Codec G.711 G.711 G.729A G.729A
Packetization interval 10 ms 20 ms 10 ms 20 ms
Overhead bytes per packet 82 82 82 82
Voice bytes per packet 80 160 10 20
Total bytes per packet 162 242 92 102
Bytes per second 16’200 12’100 9’200 5’100

Bits per second 129’600 96’800 73’600 40’800
The required bandwidth can be reduced by enabling silence suppression

(refer to section 5.4.1.9 "Silence suppression" (on page 47)).

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5.4.1.13 RTP port range
The RTP port range defines the beginning and the end of the RTP UDP port
numbers that are used for media connections. The range, i.e. “Stop Port
Number” - “Start Port Number”, should be at least 4000. The setting of the
following parameters is required:
• Start port number
• Stop port number
5.4.1.14 Real-time transport control protocol (RTCP)
RTCP is a companion protocol to RTP, used for gathering statistical informa-

tion for a media connection. Collected information can be
• bytes sent,
• packets sent,
• lost packets,
• jitter,
• round trip delay.
Every 5 seconds, IPSS4 sends a sender report (SR) packet together with a
source descriptor (SDES) packet towards the IP network. The opposite gate-
way sends then also a sender report or a receiver report packet back to
IPSS4. From this received RTCP packet IPSS4 is able to calculate the round
trip delay.
Other RTCP parameters than round trip delay are currently not evaluated by
IPSS4.
Please note:
The display of the round trip delay is supported in a future release.
Please note:
The gateway on the opposite side is able to evaluate and present the RTCP
parameters.
5.4.2 SIP
For a generic description of the SIP protocol please refer to section 5.2 "SIP
protocol" (on page 28).
In the SIP environment IPSS4 plays the role of a user agent (UA) for every
connected channel. The SIP part of IPSS4 can be configured to accommo-
date the SIP behaviour and the connection of the GW to the softswitch.
The parameters in the following sections are configurable.
5.4.2.1 Gateway name
A name of the gateway with up to 63 characters has to be configured. This

name is allocated to the GW and is used as 'username' in the “From:”, “To:”
and “Contact:” headers of the proxy availability check messages.

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IPSS4
5.4.2.2 Home domain
The home domain is the service provider domain for a SIP user. Typically,
this is the domain present in the URI in the “public address” of a registration.
The public address is in the same domain as the location service that can
map a received URI to another URI where the user might be available.
5.4.2.3 SIP port number
The SIP port number defines the SIP protocol UDP port number of the GW.
The default value is 5060.
5.4.2.4 Country code
This is the international country code for calling a subscriber connected to

the GW. The country code is required when receiving global subscriber num-
bers in SIP messages.
On receiving a number preceded with a “+”, IPSS4 searches for a match
with the “country code + area code” string. The subscriber ID is the received
number minus the “country code + area code”.
The country code is always prefixed with “+”. The default country code is
+41.
If no country code is required this field might be left empty.
5.4.2.5 Area code
This is the national area or city code (excluding the leading zero) for calling a
subscriber connected to the GW. The area code is required when receiving
global or area subscriber numbers in SIP messages.
On receiving a number preceded with a “0” IPSS4 searches for a match with
the “0 + area code” string. The subscriber ID is the received number minus
the “0 + area code”.
The default area code is 31.
If no area code is required this field might be left empty.
5.4.2.6 Retransmissions
The retransmissions configuration comprises the timer parameters T1 and

T2.
The timer value T1 is the basic retransmit timer for SIP requests and INVITE
responses.
The behaviour with INVITE requests is as follows: If after T1 the user agent
client receives no response, the INVITE request is retransmitted. The T1
timer value doubles after every retransmission. T1 is an estimate for the
round trip time. The default value for T1 is 500 ms.
The timer value T2 is the maximum retransmit interval value for non-INVITE
requests and INVITE responses. If after T1 the user agent client receives no
final response (200-699) on a non-INVITE request, the request is retransmit-
ted. The T1 timer value doubles after every retransmission up to the T2 limit.

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IPSS4
If a provisional response (1xx) is received, retransmissions continue, but with

the maximum interval of T2. The default value for T2 is 4 s.
The figure below shows the retransmissions of INVITE and non-INVITE
requests when no response is received. The retransmission “0” is the initial
transmission of the request, the retransmission “1” is the first real retransmis-
sion after the interval T1.
In this example T1 is 1 s and T2 is 8 s.
INVITE
retransmissions
0 1 2 3 4 5 6
0 1 3 7 15 31 63 time
[T1]
non-INVITE
retransmissions
0 1 2 3 4 5 6 7 8 9 10
T2
0 1 3 7 15 23 31 39 47 55 63 time
[T1]
T2 = 8 * T1
Figure 22: Request retransmissions
Note that the transaction time-out for INVITE and non-INVITE requests is
64 * T1. The maximum number of INVITE retransmissions is therefore 6 and
the maximum number of non-INVITE retransmissions is 10, if T2 = 8 * T1.
5.4.2.7 Reliable provisional response messages
In SIP only the request messages and final response messages (2xx - 6xx)
are delivered reliably. The provisional response messages (1xx) are deliv-
ered unreliably. Sometimes it is important to have reliable delivery of provi-
sional response messages as e.g. in interoperability scenarios with the
PSTN.
RFC3262 specifies an extension to SIP for reliable delivery of provisional
response messages. This extension uses the option tag “100rel”. The used
method is analogous to the method for the INVITE message delivery where
the final response is confirmed with an ACK message. In the reliable provi-
sional response message delivery the messages are confirmed with a
PRACK (PRovisional ACK) message.
Since PRACK is a normal SIP message it has its own response message.
5.4.2.8 Asserted identity and privacy
RFC3325 describes extensions to SIP that enable a network of trusted SIP

servers to assert the identity of end users or end systems, and to convey
indications of end-user requested privacy. The use of these extensions is
only applicable inside an administrative domain with previously agreed-upon
policies for generation, transport and usage of such information.
The mechanism proposed in RFC3325 relies on two new header fields and a
new privacy header type:

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IPSS4
• P-Asserted-Identity header, contains a URI and an optional display name.

If a proxy receives a message (request or response) from a node that it
trusts, it can use the information in the P-Asserted-Identity header field as
if it had authenticated the user itself.
When a proxy forwards a message to another node, it must first deter-
mine if it trusts that node or not. If it trusts the node, the proxy does not
remove any P-Asserted-Identity header fields that it generated itself, or
that it received from a trusted source. If it does not trust the element, then
the proxy examines the Privacy header field to determine if the user
requested that asserted identity information be kept private.
• P-Preferred-Identity header, contains a URI and an optional display
name.
The P-Preferred-Identity header field is used from a user agent to a
trusted proxy to carry the identity the user wishes to be used for the P-
Asserted-Header field value that the trusted element will insert. Proxies
will remove this header field.
User agents must not use the P-Preferred-Identity header field in a mes-
sage that is not sent directly to a proxy that is trusted by the user agent.
• Privacy header with type “id”.
The Privacy header type “id” indicates that the user would like the net-
work asserted identity to be kept private with respect to SIP entities out-
side the trusted domain with which the user authenticated. The proxy that
is about to forward a message to a proxy server or UA that it does not
trust will then remove all the P-Asserted-Identity header field values.
If the Privacy header field value is set to “none” then the proxy will not
remove the P-Asserted-Identity header fields.
The IPSS4 unit supports asserted identities with the possibilities to configure
the asserted identity mode per gateway and to configure authorisation user
names and passwords and the privacy header type per subscriber number.
• Asserted Identity Mode values:
− None:
No “P-…-Identity” header is added.
− Asserted:
Adds the “P-Asserted-Identity” header with Display Name (if available)
and Subscriber Number.
− Preferred:
Adds the “P-Preferred-Identity” header with Display Name (if availa-
ble) and Subscriber Number
Please note:
“P-…-Identity” headers are added to SIP requests only and not to responses.
The following parameters are available per subscriber number for ISDN-BA
and PSTN subscribers or per trunk hunting group.
• Subscriber Number:
The Subscriber Number is also the User Name and is always required.
• Authorisation User Name:
Typically this is the subscriber number including the area and country
codes. This field can be left empty if it is not required.
• Authorisation Password:
The Authorisation Password can be left empty if it is not required.
• Display Name:
The Display Name can be left empty.

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IPSS4
• Privacy:
The Privacy header type is configurable to one of the following properties:
− None:
No “Privacy” header is added.
− Id:
The Privacy header with type “id” is added.
Please note:
The Privacy header will only be added if the asserted identity mode is config-
ured to “Asserted” or “Preferred”.
5.4.2.9 Outgoing overlap signalling
While modern local exchanges always use en-bloc signalling, some parts of
the PSTN still use overlap signalling. En-bloc signalling consists of sending
the complete telephone number of the callee in the first signalling message.
Outgoing overlap signalling consists of sending only some digits of the cal-
lee’s number in the first signalling message. Further digits are sent in subse-
quent signalling messages.
Like modern local exchanges, SIP uses en-bloc signalling. The Request-URI
of an INVITE request always contains the whole address of the callee.
Therefore, the preferred solution for a gateway handling PSTN overlap sig-
nalling and SIP is to convert the PSTN overlap signalling into SIP en-bloc
signalling using number analysis and timers.
Although en-bloc signalling is the preferred solution, conversion of overlap to
en-bloc signalling sometimes results in unacceptable call setup delays. In
these situations, overlap signalling has to be used also in the SIP network to
minimize the call setup delay.
RFC3578 describes two approaches for the outgoing overlap signalling han-
dling. The IPSS4 unit supports the “multiple INVITEs” approach.
In this scenario, the gateway receives the initial digits of a dialled number via
DTMF (POTS) or a DSS2 INFO message (ISDN) and possibly one or more
subsequent digit(s). As soon as the minimum amount of digits is received,
the gateway sends an INVITE and starts the T10 timer. If subsequent digits
arrive to the gateway, the T10 timer is refreshed and a new INVITE with the
previous digits and the new digits received is sent.

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IPSS4
PSTN Gateway SIP
initial digits
Start T10
INVITE
subsequent digits
Start T10
INVITE
subsequent digits
Start T10
INVITE
Figure 23: Outgoing overlap signalling
The digit collection process is stopped as soon as a 18x (except 183) or 200
response has been received or the T10 timer has expired.
The call is established when the gateway receives a 200 OK response for
the (latest) INVITE request.
If the call can not be established, i.e. if 4xx, 5xx or 6xx final responses arrive
(e.g. 484 address incomplete) for all INVITEs sent, and the T10 timer has
expired, the gateway will release the call setup and play the congestion tone.
PSTN Gateway SIP
initial or subsequent digits

Start T10
INVITE
484 address incomplete
ACK
T10 expires
REL (release message)
Figure 24: Call release with expiry of the T10 timer
Please note:
The “multiple INVITEs” approach works only reliably if all the INVITE mes-
sages reach only one gateway. The network administrator must therefore
ensure the correct routing of the SIP messages.
Outgoing overlap signalling is available for POTS and for ISDN-BA subscrib-
ers.
Following parameters regarding overlap signalling are provisioned by the
ECST in the IPSS4 unit configuration AP: /unit-x, Configuration - Gateway -
SIP:

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IPSS4
• Overlap Signalling (true/false).

• Overlap T10 Timer (1 … 60 s).
The number of digits contained in the initial INVITE has to be configured via
an appropriate digitmap.
Please note:
The digitmap shall be carefully specified as it might cause a malfunction of
the overlap signalling feature, e.g. if the digitmap ends with a dot (“.”), then
the initial INVITE message is not sent immediately but after the expiry of the
short timer.
5.4.2.10 Incoming overlap signalling
Incoming overlap signalling is used in call establishment of an incoming call,

when the network sends the called party number digits to the user in succes-
sive messages.
ETSI TS 183 036 describes two approaches for the incoming overlap signal-
ling handling of direct dialling in (DDI). The IPSS4 unit supports the “multiple
INVITEs” approach.
When the IPSS4 receives the first INVITE for a subscriber connected to a
PABX, it sends a SETUP message containing all the digits received in the
INVITE message.
The PABX needs more digits to route the call and sends a SETUP
ACKNOWLEDGE to the IPSS4 indicating the insufficient number of digits.
The IPSS4 enters now the “incoming overlap signalling” state, sends a SIP
484 (address incomplete) response and starts the timer T304.
The IPSS4 will send an INFORMATION message upon receipt of each sub-
sequent INVITE message with the same Call-ID and From tag as the first
INVITE message. The INFORMATION message contains the additional dig-
its received in the INVITE message.
Once the PABX has all the digits required it sends a CALL PROCEEDING
message and the call will be setup.
With each sending of an INFORMATION message the timer T304 restarts.
The timer stops when the CALL PROCEEDING message is received.
Upon expiry of T304 the call is cleared. The timer T304 has a fixed value of
20 s.

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IPSS4
SIP Gateway PABX
Timer T304
INVITE (90012) [seq 1]
SETUP (90012)
SETUP ACKNOWLEDGE
Start
INVITE (9001201) [seq 2]
INFORMATION (01)
Restart
484 (address incomplete) [seq 1]
ACKNOWLEDGE [seq 1]
INVITE (90012014) [seq 3]

INFORMATION (4)
Restart
484 (address incomplete) [seq 2]
ACKNOWLEDGE [seq 2]
CALL PROCEEDING
Stop
183 (progress) [seq 3]
Figure 25: Incoming overlap signalling
Please note:
Incoming overlap signalling is available for ISDN-BA subscribers of trunk
hunting groups only.
IPSS4 provides no configuration parameters for the incoming overlap signal-
ling.
5.4.2.11 Session timers
Session timers are an extension to the SIP protocol and are defined in
RFC4028. This extension allows for a periodic refresh of SIP sessions
through a re-INVITE or UPDATE request. The refresh allows both user
agents and proxies to determine whether the SIP session is still active.
Two new header fields and a new response code are defined for the session
timer feature:
• Session-Expires header:
Conveys the lifetime of the session with the “session interval” parameter.
This is the upper bound for the session refresh interval; i.e. the time
period for which any session-stateful proxy must retain its state for this
session. Any proxy or UAS servicing this request can lower this value, but
it is not allowed to decrease it below the value specified in the Min-SE
header field.
The Session-Expires header field also contains a “refresher” parameter,
which indicates who is doing the refreshing
− the UA that is currently the UAC, or
− the UA that is currently the UAS.

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• Min-SE header:
Conveys the minimum allowed value, called “minimum timer” for the “ses-
sion interval”.
Because of the processing load of mid-dialogue requests, all elements
(proxy, UAC, UAS) can have a configured minimum value for the session
interval that they are willing to accept. Each proxy processing the request
can raise this lower bound but is not allowed to lower it.
• 422 response code:
Indicates that the proposed session interval was too small.
If this is the case for a proxy (i.e., the “session interval” is lower than the
“minimum timer” that the proxy would wish to assert), the proxy rejects
the request with a 422 response. That response contains a Min-SE
header field identifying the minimum session interval it is willing to sup-
port.
From the Session-Expires header field in the response of the UAS to the ini-
tial INVITE, both user agents know that a session timer is active, when it will
expire, and who is refreshing. At some point before the expiration, the cur-
rently active refresher generates a session refresh request, which is a re-
INVITE or UPDATE request. The user agent generating the 2xx response to
the re-INVITE or UPDATE request extends its session expiration time by the
session interval time. The user agent receiving the 2xx response also
extends its session expiration time by the session interval time.
If the refresher never gets a response to that session refresh request, it
sends a BYE to terminate the session. Similarly, if the other side never gets
the session refresh request before the session expires, it sends a BYE.
The IPSS4 unit implements session timers with the following parameters:
• IPSS4 includes the “timer” tag in the “Supported:” header to indicate that
the UA supports the SIP session timer.
• IPSS4 does not specify the “refresher” when initiating a call. The
refresher is specified/negotiated in the 2xx responses.
• IPSS4 uses the INVITE request for the session refresh. The UPDATE
request is not used.
• The “Session Expiration” configuration parameter is the “session interval”
used in the Session-Expires header. The value is configurable from 180 s
(3 min) to 86’400 s (24 h), with a default value of 1’800 s (30 min).
• The default value of the “refresher” used in the Session-Expires header is
fixed to UAS.
• The default value of the “minimum timer” used in the Min-SE header is
fixed to 90 s.
• When the configuration parameter “UAC Request Session-Timer” is ena-
bled IPSS4 includes the Session-Expires and the Min-SE headers in the
INVITE request.
If the 2xx response to this INVITE request does not contain the Session-
Expires headers, i.e. if the other side does not support session timers,
IPSS4 will not make use of the session timers.
When “UAC Request Session-Timer” is disabled IPSS4 does not include
the Session-Expires and the Min-SE headers in the INVITE request.
• When the configuration parameter “UAS Request Session-Timer” is ena-
bled IPSS4 checks whether the “timer” tag is present in the SIP “Sup-
ported” header of the initial INVITE request. IPSS4 inserts in the 2xx
response the Session-Expires and the Min-SE headers and will perform
session refresh.

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IPSS4
If the initial INVITE request does not contain the “timer” tag, i.e. if the
other side does not support session timers, IPSS4 will not make use of
the session timers.
When “UAS Request Session-Timer” is disabled IPSS4 does not include
the Session-Expires and the Min-SE headers in the 2xx response to the
INVITE request.
5.4.3 Supplementary services implemented for PSTN subscribers
The following invocation codes for the supplementary services of PSTN sub-
scribers are implemented in the IPSS4 gateway.
The invocation codes are applied by the PSTN subscriber by pressing the
hook-flash button (R-button), wait for the recall dial tone (rdt) and then dial-
ling the number of the desired action.
The invocation codes are configurable in the gateway. The values in Table
13 are the default values.
Table 13: Default invocation codes
Service Action
(R = hook flash)
Reject a waiting call R+0
Disconnect an active call R+1
Toggle to a call on hold or accept a waiting call R+2
3-way call between a call on hold and an active call R+3
Transfer a call on hold to an active call R+4
IP
Gateway
Network
C
Figure 26: Call scenario with three participants
Supplementary services can be implemented on the gateway or on the softs-

witch:
Table 14: Supplementary service implementation
Supplementary service Implemented on
gateway softswitch
Hotline (HOTL) 
Per call calling line ID restriction (CLIR) 
Calling line ID presentation (CLIP) 
Call waiting (CWT/CWI)  
Enquiry call (ENQ)  

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Table 14: Supplementary service implementation (continued)

Supplementary service Implemented on
gateway softswitch
3-way call (3WC)  
Attended call transfer (CXR)  
Call forwarding unconditional (CFU) 
Call forwarding busy (CFB) 
Call forwarding no reply (CFnR) 
Services that are implemented on the softswitch require the gateway to for-
ward feature codes which where dialled by the subscriber to the softswitch.
Feature codes are passed as request URI (uniform resource identifier) to the
softswitch. The network operator has to define a digit map with the appropri-
ate domain/phone context to support the forwarding of the feature codes to
the appropriate destination.
Please note:
Any codes for supplementary services on the softswitch require the configu-
ration of a corresponding digit map entry.
The codes of predefined supplementary services in the ECST (e.g. *31*)
have a predefined (invisible) digit map entry.
5.4.3.1 Call waiting
With the call waiting supplementary service, the user is, during an active call,
informed with an appropriate indication (call waiting tone) that there is
another incoming call waiting for him to be answered.
The call waiting supplementary service can be activated and deactivated by
the subscriber. The corresponding codes are configurable in the gateway.
Furthermore it is possible to have the call waiting supplementary service ini-
tially activated or deactivated, i.e. when a port is added to a portgroup or the
port configuration is changed.
Table 15: Call waiting call scenario
Action / notification Call status
Subscriber A Subscriber B Subscriber C
1 accept call call A call established
2 notified by call call A between A and B
waiting tone
3 apply hook flash B on hold
accept (alternative 1)
41 press 2 (accept) call established
between A and C
51 apply hook flash C on hold
61 press 2 (toggle) call established
between A and B
reject (alternative 2)
42 press 0 (reject) notified about A call established
busy between A and B

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5.4.3.2 Enquiry call
With the enquiry call supplementary service, the user can, during an active
call, initiate a second call to another subscriber for e.g. an enquiry, terminate
this call and then go back to the initial call.
Table 16: Enquiry call scenario
between A and B
3 call C accept call call established
between A and C
disconnect (alternative 1)
51 press 1 (discon- call between A
nect) and C terminated;
call established
between A and B
toggle (alternative 2)
52 press 2 (toggle) call established
between A and B
on hook / off hook (alternative 3)
43 go on hook call between A
and C terminated;
ringing is applied
at A to notify the
call to B being on
hold
53 go off hook call established
between A and B
5.4.3.3 3-Way call
With the 3-way call supplementary service, the user can, during an active
call, initiate a second call to another subscriber to establish a 3-way confer-
ence call.
Table 17: 3-way call scenario
between A and B
between A and C

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IPSS4
Table 17: 3-way call scenario (continued)

5 press 3 (3-way call established
call) between A, B and
C
hook flash (alternative 1)
61 apply hook flash no action
A goes on hook (alternative 2)
62 go on hook all calls are termi-
nated
B goes on hook (alternative 3)
63 go on hook call established
between A and C
C goes on hook (alternative 4)
64 go on hook call established
between A and B
Please note:
A subscriber that initiated a 3-way call cannot participate in another 3-way
call initiated from another subscriber on the same gateway.
This limitation is not applicable if the involved subscribers are located on dif-
ferent gateways or if a session border controller (SBC) is used as a back-to-
back user agent.
5.4.3.4 Attended call transfer
With the call transfer supplementary service, the user can, during an active
call, initiate a second call to another subscriber to transfer the initial call to
the newly established call.
Table 18: Attended call transfer scenario
between A and B
between A and C
5 press 4 (transfer) call established
between B and C
61 successful trans- call between A
fer is notified by and B terminated;
confirmation tone call between A
and C terminated
62 failed transfer is
notified by conges-
tion tone

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Table 18: Attended call transfer scenario (continued)

63 hook flash after call established
failed transfer between A and B
5.4.3.5 Calling line ID presentation (CLIP)
The calling line identity presentation (CLIP) supplementary service makes it

possible to display the identity of the calling user on the subscribers terminal.
The CLIP supplementary service uses FSK signalling or DTMF signalling for
the data transport. The corresponding configuration is done in the unit con-
figuration or the PSTN port profile.
IPSS4 supports the following sequences:
− Data transmission during ringing (FSK)
− Data transfer during off-hook (FSK or DTMF)
The usage of SIP headers for CLIP and display information is controlled by
CPS parameters as follows:
• The CPS parameter DEV1.5 controls if CLIP displays the “User Number”
or the “Display Name”.
• The CPS parameter SRV1.10 controls the mapping of the SIP headers
“P-Asserted-Identity” and “From:” to the CLIP information for PSTN sub-
scribers. The default is as follows:
− If the “P-Asserted-Identity” header is present, CLIP will take the dis-
play information from this header, otherwise from the “From:” header.
• The CPS parameters SRV1.11 and SRV1.13 control the mapping of the
SIP headers “P-Asserted-Identity” and “From:” to the first and second
“Calling Party Number” Information Element (IE) for ISDN-BA subscrib-
ers. A second “Calling Party Number” IE is only inserted if the “P-
Asserted-Identity” and “From:” header contents are different.
The default is as follows:
− If the “P-Asserted-Identity” header is present, the first “Calling Party
Number” IE will take the display information from this header, other-
wise from the “From:” header. No second “Calling Party Number” is
sent.
• The CPS parameters SRV1.12 and SRV1.14 control the screening indi-
cator of the first and second “Calling Party Number” IE for ISDN-BA sub-
scribers. The screening indicator can be set to
− user provided, not screened,
− network provided, or
− user provided, verified and passed.
The default is as follows:
− for the first “Calling Party Number” IE: user provided, not screened,
− for the second “Calling Party Number” IE: network provided.
• The CPS parameter SRV1.15 controls the mapping of the “Calling Party
Number” and of the registered “Default Number” to the SIP headers “P-
Preferred-Identity” or “P-Asserted-Identity” and “From:”. This applies for
huntgroups only with the “Special Arrangement” configured.

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Please note:
The selection between the “P-Preferred-Identity” and the “P-Asserted-Iden-
tity” headers is done with the IPSS4 unit configuration parameter “Asserted
Identity Mode”.
• CLIP makes also use of the Calling Line Identity parameters. IPSS4 will
send the information from the “Request-URI” or “To:” header, controlled
by the CPS parameter DEV1.6.
In case the “P-Called-Party-ID” header is present, IPSS4 will send the
information from this header. The “P-Called-Party-ID” header applies only
for huntgroups.
Please note:
For a description of the CPS please refer to section 6.1 "Customer profiles"
(on page 93).
5.4.3.6 Hotline
The hotline (HOTL) supplementary service allows the subscriber to automati-

cally dial a preconfigured number after a configurable time-out.
The supplementary service can be activated and deactivated by the sub-
scriber. The activation or deactivation state can be interrogated by the sub-
scriber. The corresponding codes are configurable in the gateway.
Activation call scenario example:
− Go off hook
− Activate the hotline service: Dial *53*
− Dial the subscriber number to be called
− Terminate the subscriber number with the # sign
− You receive the confirmation message or a confirmation tone
− Go on hook
Deactivation call scenario example:
− Go off hook
− Deactivate the hotline service: Dial #53#
− You receive the confirmation message or a confirmation tone
− Go on hook
Interrogation call scenario example:
− Go off hook
− Interrogate the activation state of the hotline service: Dial *#53#
− You receive the activation or deactivation message or a confirmation
tone
− Go on hook
5.4.3.7 Per call calling line ID restriction (CLIR)
The calling line identity restriction (CLIR) supplementary service allows the
subscriber to suppress the display of his identity at the called subscriber’s
terminal on a per call basis.
The supplementary service can be activated by the subscriber. The corre-
sponding codes are configurable in the gateway.
Call scenario example:

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IPSS4
− Go off hook
− Activate the CLIR service: Dial *31*
− Dial the called subscriber number
− Terminate the called subscriber number with the # sign
Please note:
This supplementary service is implemented on the softswitch. It is only avail-
able if the service is supported by the softswitch!
The behaviour of this service might be adapted via CPS parameters, e.g.
removing the activation code and/or the terminating character and use pri-
vacy values (id, header, user) in the SIP message to enable 'Per Call CLIR'.
5.4.3.8 Call forwarding
The IPSS4 supports the following three call forwarding supplementary ser-
vices (Note that the respective call forwarding service implementation has to
be available on the softswitch):
• CFU: Call forwarding unconditional allows the subscriber to forward any
incoming call to another subscriber number.
• CFB: Call forwarding busy allows the subscriber to forward an incoming
call to another subscriber number if he is busy, i.e. off hook.
• CFnR: Call forwarding no reply allows the subscriber to forward any
incoming call to another subscriber number if the call is not answered in a
(proxy) predefined time.
IPSS4 supports the call forwarding supplementary services for
• PSTN subscribers
− via a keypad protocol.
• ISDN-BA subscribers
− via a keypad protocol, or
− via the generic functional protocol (GFP).
The supplementary services can be enabled or disabled per gateway or per
subscriber.
If the supplementary services are enabled they can be activated and deacti-
vated by the subscriber. The activation or deactivation state can be interro-
gated by the subscriber. The corresponding keypad codes have to be config-
ured in the gateway.
ISDN-BA subscribers can use the diversion interrogation GFP command to
get the supplementary status information. In this case, the status information
on Call Forwarding will be received from the local database on the IPSS4
and not from the network. Furthermore, the information will only be available
in the local database if GFP was used to activate or deactivate the Call For-
warding services.
Service activation example:
− Go off hook
− Activate the CFxx service: Dial *21*
− Dial the called subscriber number
− Terminate the called subscriber number with the # sign
− Go on hook
Service deactivation example:
− Go off hook

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− Deactivate the CFxx service: Dial #21#

− Go on hook
Service interrogation example:
− Go off hook
− Interrogate the CFxx service: Dial *#21#
− Go on hook
For ISDN-BA subscribers the GFP controlled Call Forwarding supplementary
service status is available in the ECST status management function of the
ISDN-BA port.
5.4.3.9 Message summary and message waiting indication (MWI)
IPSS4 supports the message summary and message waiting service

according to IETF RFC3842, without usage of the SUBSCRIBE portion of
this RFC.
Any available voice message is signalled to the gateway with a NOTIFY
message. After receiving a NOFIFY message with the voice message indica-
tion, an FSK sequence is sent to the subscriber’s telephone set immediately.
The FSK sequence contains the following information:
− Calling line identity, i.e. the voice mail box subscriber number,
− Visual indicator,
− Number of messages,
− Type of calling user.
Additionally the gateway provides stutter dial tone instead of regular dial tone
to a subscriber going off hook.
5.4.3.10 Multiple subscriber numbers (MSN)
Multiple subscriber numbers (MSN) enables end-users to have up to 3 indi-

vidual numbers on a single PSTN subscriber port. The first number in the list
of MSN numbers becomes the default PSTN number. This number will be
registered as such in the network for identification purposes (default PSTN
number for supplementary services).
The custom parameter set (CPS) provides a specific cadence ringing type
for each subscriber number.
5.4.3.11 Subscriber time correction
Subscriber time correction allows you to adjust the time delivered to the end
equipment according to the local daylight saving policy.
The IPSS4 supports two daylight saving policies:
• European standard policy:
− time offset is 1 hour,
− start date and time is the last Sunday of March, at 01h00 AM,
− end date and time is the last Sunday of October, at 01h00 AM.
• Non-European standard policy:
− time offset is configurable up to 12 hours,
− start and end date and time are configurable.

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5.4.4 Enterprise supplementary services implemented for PSTN subscribers
The following supplementary services for PSTN subscribers are controlled

by the SIP enterprise profile, applicable to the PSTN port configuration (AP: /
unit-x/portgroup-y/port-z, Configuration). Alternatively the unit configuration
(AP: /unit-x, Configuration - Gateway - Supplementary Services - Available
Services) of the enterprise supplementary services can be applied to all
PSTN ports which have set the “Enterprise Profile” to “none”.
Please note:
Any codes for enterprise supplementary services require the configuration of
a corresponding digit map entry.
5.4.4.1 External dial tone
With the external dial tone supplementary service, the user is, during dialling
an external number, informed with an appropriate indication (specific exter-
nal dial tone) that he or she is in the course of dialling an external number.
The external dial tone is defined within the CPS.
The prefix of the external number is configurable with up to three digits with
the “off-net prefix” parameter.
The procedure is as follows:
• After going off-hook the gateway plays the normal dial tone towards the
subscriber.
• The subscriber dials the first digit, the dial tone is stopped.
• When the subscriber has finished dialling the digits matching the config-
ured “Off-Net Prefix” the gateway plays the external dial tone.
• The subscriber dials the next digit, the external dial tone is stopped.
Please note:
This supplementary service is disabled when the “off-net prefix” field is
empty.
Please note:
The shortest provisioned digit map entry must be equal or longer than the
“off-net prefix”, otherwise the external dial tone feature is not going to work.
5.4.4.2 Blocked number prefixes
With the blocked number prefixes supplementary service, all or some PSTN
subscribers are blocked from dialling numbers with specific prefixes, e.g.
national or international numbers.
A maximum of 3 different blocked prefix sequences may be configured with
a maximum length of 5 digits.
If a subscriber dials a blocked number the gateway plays the congestion
tone towards the subscriber.
Please note:
This supplementary service is disabled when the “blocked number prefixes”
table is empty.

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Please note:
The shortest provisioned digit map entry must be equal or longer than the
“blocked prefix”, otherwise the blocked prefix feature is not going to work.
5.4.4.3 Hotline
The enterprise hotline supplementary service allows an operator to configure

a hotline number. The hotline number is automatically dialled if a subscriber
goes off-hook and the hotline start timer has elapsed.
The hotline start timer is configurable in the range from 0 to 9 seconds.
When the hotline start timer is configured to 0 seconds the dialling from
PSTN telephone sets is completely blocked.
Please note:
The enterprise hotline supplementary service has precedence over the
PSTN subscriber hotline supplementary service.
→ If the enterprise hotline supplementary service is enabled a PSTN
subscriber is no longer able to enable, disable or interrogate the sub-
scriber hotline supplementary service.
5.4.5 Supplementary services implemented for ISDN-BA subscribers
IPSS4 supports the same supplementary services for ISDN-BA subscribers

as for PSTN subscribers. Please refer to section 5.4.3 "Supplementary ser-
vices implemented for PSTN subscribers" (on page 60).
Please note:
ISDN-BA subscribers can use generic functional protocol (GFP) messages
for the support of supplementary services.
In addition to the supplementary services listed for the PSTN subscribers,
the following supplementary services are available for ISDN-BA subscribers:
• Multiple Subscriber Numbers (MSN)
• Call Deflection (CD)
• Partial Rerouting (PR)
• Terminal Portability (TP)
• Direct Dialling In (DDI)
• Trunk Hunting (THU)
• Advice Of Charge (AOC)
5.4.5.1 Multiple subscriber numbers (MSN)
Multiple subscriber numbers (MSN) enables end-users to have up to 31 indi-

vidual numbers on a single basic access. The first number in the list of MSN
numbers becomes the default ISDN number. This number will be registered
as such in the network for identification purposes (default ISDN number for
supplementary services).

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5.4.5.2 Call deflection (CD)
Call deflection (CD) permits a subscriber to request that the network redi-
rects an incoming call addressed to the subscriber's ISDN number to
another number (the deflected-to-number).
The subscriber sends the network a deflection request with the deflected-to-
number provided in the request. The subscriber's originating service is unaf-
fected.
Unlike “call forwarding”, the network will redirect a call only after receipt of a
specific user request to deflect that call.
5.4.5.3 Partial rerouting (PR)
PR is used together with the call forwarding and call deflection supplemen-
tary services.
If an incoming call from the public ISDN to e.g. a PABX has to be rerouted to
a destination in the public ISDN, the call is rerouted in the public ISDN. The
resources in the IPSS4 and the subscriber lines are freed.
5.4.5.4 Terminal portability (TP)
Terminal portability (TP) enables a subscriber engaged in an active call to

adjourn communication by an appropriate signalling procedure and resume
the call at a later time.
When a call is successfully suspended or resumed, a notification is sent to
the other user. This procedure applies independently to calling and called
users.
The TP supplementary service allows a subscriber during the active phase
of a call
− to replace one terminal by another compatible terminal at the same
socket.
− to move from one terminal to another compatible terminal within a
basic access.
− to suspend the call and subsequently to resume it at the same termi-
nal and the same socket.
− to move a terminal from one socket to another one.
The portability of a terminal in the call establishment and in the call clearing
phases is not possible.
5.4.5.5 Direct dialling in (DDI)
For the description of the direct dialling in (DDI) supplementary service

please refer to section 5.6 "Direct dialling in and trunk hunting" (on page 86).
5.4.5.6 Trunk hunting (THU)
For the description of the trunk hunting (THU) supplementary service please
refer to section 5.6 "Direct dialling in and trunk hunting" (on page 86).

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5.4.5.7 Advice of charge (AOC)
Advice of charge (AOC) enables a subscriber to receive information about

the charging rates of the call.
The IPSS4 supports the following three variants of AOC:
• Charging information at call set-up time (AOC-S):
The AOC-S supplementary service enables a subscriber to receive infor-
mation about the charging rates at call set-up time and also to receive
further information during the call if there is a change of charging rates.
• Charging information during the call (AOC-D):
The AOC-D supplementary service enables a subscriber to receive infor-
mation on the recorded charges for a call during the active phase of the
call.
• Charging information at the end of the call (AOC-E):
The AOC-E supplementary service enables a subscriber to receive infor-
mation on the recorded charges for a call when the call is terminated.
5.4.6 Network interface functions
The network interface functions implement the interface towards the IP net-
work. The parameters described in the following sections have to be config-
ured.
5.4.6.1 Interface parameters
IPSS4 requires the configuration of the

• Gateway IP address
This parameter defines the IPv4 address of the gateway.
• Subnet mask.
• Default gateway
This parameter defines the IPv4 gateway address.
There are no default values for the above parameters.
5.4.6.2 QoS parameters
IPSS4 supports differentiated services (DiffServ) as a method to guarantee

quality of service (QoS) on large networks.
With differentiated services traffic is divided into a certain number of forward-
ing classes, and resources are allocated on a per-class basis. The DSCP
(Differentiated Service Code Point) is a number in the range of 0 … 63. It
uses 6 bits in the ToS (type of service) byte in the IP header.
The DSCP has to be configured for control and media traffic:
• DSCP for SIP control traffic.
• DSCP for voice, voice band data (VBD) and DTMF relay service
(RFC2833) traffic.
• DSCP for clearmode traffic.
• DSCP for fax relay service (T.38) traffic.

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IPSS4
The default DSCP for SIP control, clearmode and fax relay service is 40
(= class selector 5). The default DSCP for voice, VBD and DTMF relay is 46
(= expedited forwarding).
5.4.6.3 VLAN for the network interface
A network interface is assigned to a VLAN. The service VLAN ID is config-

ured at the AP: /unit-x in the VLAN tab of the gateway configuration. The
VLAN priority can be set from CoS 0 to CoS 7. It is recommended to give the
voice packets a VLAN priority of CoS 6.
The network interface of the IPSS4 unit is connected to a MileGate chassis
switch port via the double GbE star, i.e. to an internal Ethernet port on the
COGE5 unit.
E.g. a IPSS4 unit plugged in slot-8 of the MileGate subrack accesses the
internal COGE5 ports at the APs /unit-11/iports/iport-8 (working COGE5 unit)
and /unit-13/iports/iport-8 (protecting COGE5 unit).
Please note:
On the IPSS4 unit only the Ethernet link to the active COGE5 unit (working
or protecting) is active. The Ethernet link to the standby COGE5 unit (work-
ing or protecting) is inactive.
The internal chassis switch ports on the working and on the protecting
COGE5 unit must be configured with the same parameter values:
• Bridge Port Mode.
Select a bridge port mode supporting tagged traffic:
− General (preferred):
The General port mode forwards only the VLAN tagged traffic with the
assigned VLAN ID.
− Trunk:
The Trunk port mode forwards all unicast VLAN tagged traffic with the
learned MAC addresses and all multicast and broadcast VLAN tagged
traffic. This can lead to an overload situation of the traffic handling
resources on the IPSS4 unit.
• Bridge Port Acceptable Frametypes.
Set the acceptable frame types on the chassis switch port to “Tagged”.
• Bridge VLAN.
− Create a bridge VLAN with the VLAN ID matching the configured
VLAN ID of the IPSS4 network interface.
− This VLAN has to be assigned to the chassis switch port which is con-
figured to the “General” port mode.
− The VLAN is automatically assigned to the chassis switch port which
is configured to the “Trunk” port mode.
In addition to the internal COGE5 chassis switch ports also the uplink chas-
sis switch port has to be configured. The uplink chassis switch port must be
able to forward VLAN tagged traffic with the configured VLAN ID of the
IPSS4 network interface.
Please refer to [341] Quick Guide “Ethernet Switching” for a detailed descrip-
tion of the chassis switch port configuration.

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5.4.7 Digit maps
A digit map is a dialling plan resident in the GW used for detecting and
reporting digit events received on a termination.
Up to 16 digit map strings can be configured in the IPSS4 unit for the
DTMF digit collection. Each digit map string stands for up to 8 valid sub-
scriber numbering scheme alternatives. For each numbering scheme the
domain or phone context, i.e. the domain address of the telephone service
provider, and the URI (uniform resource identifier) schema can be config-
ured. This allows the operator to configure a numbering scheme with a pro-
vider domain for the national calls and a different numbering scheme and
provider domain for the international calls.
Please note:
If the “Domain / Phone Context” field is left empty then the provisioned
“Home Domain” is applied as “Domain / Phone Context”.
→ This behaviour is required when working with different SIP profiles, i.e.
different Home Domains.
The URI schema is added to the SIP INVITE request in the INVITE header
and the To: header field. It can be configured to sip or tel.
• Sending side:
− If the schema is “sip” then the URI is based on calledPartyNum-
ber@calledPartyDomain.
(e.g. sip: 3771024@KEYMILE.com)
− The “tel” schema is used only for the called party (Request-URI and
To:). The calling party always uses the sip schema (Contact: and
From:).
− If the schema is “tel” and an international number is dialled (starting
with 00) then no phone-context is added (e.g. tel: 0041313771024).
− If the schema is “tel” and no international number is dialled (not start-
ing with 00) then a phone-context is added with the information given
in the phone-context (e.g. tel: 3771024; phone-context = KEY-
MILE.com).
• Receiving side:
− The receiving side uses the country and area code configuration (unit
configuration: gateway - SIP).
Note that the country code is always prefixed with “+” and the area
code must not be prefixed with “0”.
− On receiving a number preceded with a “+” IPSS4 searches for a
match with the “country code + area code” string. The subscriber ID is
the received number minus the “country code + area code”.
− On receiving a number preceded with a “0” IPSS4 searches for a
match with the “0 + area code” string. The subscriber ID is the
received number minus the “0 + area code”.
− On receiving a number starting with a digit which is not “0”, IPSS4
considers this as a local number which is also the subscriber ID.
IPSS4 supports the following called party number manipulation per digitmap:
• Pre Strip:
Strip the first 0 … 16 digits from the called party number before any addi-
tional processing is done.
• Pre Pend:

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IPSS4
After a potential “pre strip” of digits, prepend the configured string of

0 … 16 digits to the called party number before actually placing the call.
The digit map entries are added in the “Digit Map” configuration dialogue.
Each digit map string is an alternative event sequence for the digit map, i.e.
an alternative numbering scheme. The digit map strings must be configured
according to the following rules:
− Valid digit map symbols are the digits 0 to 9, the star sign (*) and the
hash sign (#).
− A range of events contains the first and the last valid digit map sym-
bol that can match an event. The range is placed in squared brackets,
e.g. [2-5] matches with the digits 2, 3, 4 or 5.
− A set of alternative events contains a list of all possible valid digit
map symbols that can match an event. The set is placed in squared
brackets, e.g. [2356] matches with the digits 2, 3, 5 or 6.
− The letter x is used as a wildcard, replacing a symbol in the range 0
to 9.
− The dot symbol (.) stands for zero or more repetitions of the symbol,
range of events, set of events or wildcard that precedes it.
− The vertical bar symbol (|) stands for the delimitation between sub-
scriber numbering scheme alternatives.
− The digit map string is specified as a sequence of digit map symbols,
range of events, set of alternative events or wildcards.
The following table shows some examples for digit map strings that all can
be placed into one digit map:
Table 19: Digit map string examples
Digit map string Valid numbers
1234 1234
12[1-5]4 1214, 1224, 1234, 1244, 1254
123[13579] 1231, 1233, 1235, 1237, 1239
123x any 4 digit number starting with 123, i.e.
1230, 1231, 1232, 1233, 1234, 1235, 1236, 1237, 1238, 1239
123xxx any 6 digit number starting with 123
123x. any number starting with 123
123x.|xxxx any number starting with 123 or any 4 digit number
*xx# any two digit number preceded by a * and succeeded by a #
x. any number
Please note:
Any codes for supplementary services on the softswitch require the configu-
ration of a corresponding digit map entry.
The codes of predefined supplementary services in the ECST (e.g. *31*)
have a predefined (invisible) digit map entry.
The timers used for the digit collection process are common for all digit map
entries. There are three timers to be configured:
• Start timer:
The start timer defines the waiting time for the first digit after an off hook
event.
• Long timer:

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The long timer is an interdigit timer. It defines the waiting time for a digit
when at least one more digit is expected to complete a string in the digit
map.
• Short timer:
The short timer is an interdigit timer. It defines the waiting time for a digit
after the completion of a digit map string, if more digits could be received
to match another digit map string.
Please note:
The hash sign (#) terminates the digit collection process without waiting for
the expiry of the short timer. The terminating character (#) is not forwarded
to the proxy.
Please note:
The hash sign does not act as a terminating character if it is the first charac-
ter in a dialling sequence or if the sequence starts with a star sign (*).
If a timer expires the digit collection process is terminated. The digit collec-
tion process is terminated immediately if the match is unambiguous.
The following example shall clarify the usage of the three timers. Lets
assume that there are three event sequences (digit map strings) defined:
− Sequence 1 consists of 4 digits.
The subscriber dials a subscriber number with 6 digits:
1. Subscriber goes off hook. The start timer is activated.
2. Before the start timer expires, the first digit is received.
3. The long timer is started.
4. Before the long timer expires, the second digit is received.
6. Before the long timer expires, the third digit is received.
8. Before the long timer expires, the fourth digit is received. With the fourth
digit the event sequence 1 is completed. Further digits could complete
the event sequence 2 or 3.
9. The short timer is started.
10. Before the short timer expires, the fifth digit is received.
12. Before the long timer expires, the sixth digit is received. With the sixth
digit the event sequence 2 is completed. Further digits could complete
the event sequence 3.
13. The short timer is started.
14. The short timer expires without any received digit. The digit collection
process terminates with a “full match” for the event sequence 2.

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IPSS4
start long long long short long short

timer timer timer timer timer timer timer
Start digit 1. 2. 3. 4. 5. 6. Time out

digit digit digit digit digit digit time
collection completion
(off-hook) (full match)
digit map
event sequence 1
1. 2. 3. 4.
time
digit digit digit digit
digit map
event sequence 2
1. 2. 3. 4. 5. 6.
time
digit digit digit digit digit digit
digit map
event sequence 3
1. 2. 3. 4. 5. 6. 7. 8.
digit time
digit digit digit digit digit digit digit
Figure 27: Digit collection example
The digit map definition is according to the H.248 / MEGACO standard.

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5.5 Proxy server and registrar server functions
5.5.1 Proxy server and registrar server configuration
The SIP proxy server and registrar server have been presented up to now as
softswitch. In real applications proxy and registrar can be located at different
places and can be accessed with two different IP addresses.
The IPSS4 unit allows the operator to configure up to two proxies, a primary
and a secondary proxy.
The SIP proxy server configuration provides the following parameters:
• Proxy mode
− Primary only
All requests are sent to the primary proxy only. If the proxy fails no fur-
ther calls can be done, except local calls (if enabled via CPS).
− Non-revertive
Requests are sent to the primary proxy as long as the proxy availabil-
ity check succeeds. In case the proxy check fails on the primary proxy
then requests will be sent to the secondary proxy.
Requests are then sent to the secondary proxy as long as the proxy
availability check on the secondary proxy succeeds. In case the proxy
check fails on the secondary proxy then requests will be sent to the
primary proxy.
If the proxy availability checks on both proxies succeed or fail no
switching will occur.
− Revertive
The same behaviour as with the “Non-revertive” case applies with the
following exception:
If the proxy availability checks on both proxies succeed then requests
will be sent to the primary proxy.
Please note:
The proxy availability check must be enabled in the revertive and non-rever-
tive proxy modes.
→ If the check is disabled the proxies are assumed to be available and
no switching will occur.
− DNS / RFC3263
This mode means that no proxy has to be configured, i.e. the route is
found via DNS SRV record queries.
• Proxy address
This is the IPv4 address or the fully qualified domain name (FQDN) of the
SIP proxy server. A newly configured address causes all ongoing calls to
be cleared.
There is no default value.
Using an FQDN proxy address requires the configuration of a domain
name system (DNS) server.
• Proxy port number
This parameter defines the SIP protocol UDP port number of the SIP
proxy server. A newly configured port number causes all ongoing calls to
be cleared.

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• Proxy availability check
The gateway can check the availability of its proxy server by sending
periodically an OPTIONS or a REGISTER request. If no outbound proxy
is configured the OPTIONS or REGISTER request is sent to the registrar.
If no answer is received the proxy is assumed not to be available which
generates an alarm.
The default value for the check is true.
Please note:
To ensure reliability and reducing SIP messages sent to an unavailable
proxy/registrar, it is strongly recommended to have the proxy availability
check enabled
Please note:
If the proxy availability check is disabled, the subscriber registration process
starts as soon as the gateway unit has booted, regardless of the proxy avail-
ability.
If the proxy availability check is enabled, the subscriber registration process
starts only when the proxy is available. This may speed up the registration
process.
Please note:
If the proxy is not available, IPSS4 is still able to route local calls (LCR).
• Proxy availability check method

This parameter defines the proxy availability check is done with a
OPTIONS or with a REGISTER request.
The “From:”, “To:” and “Contact:” headers in an OPTIONS or REGISTER
request are initialized with the IPSS4 gateway name as username and
the gateway IP address as hostname.
The default value is OPTIONS.
• Proxy availability check interval
This parameter defines the interval at which the proxy availability check
shall be performed.
Note that during proxy unavailability the check interval is 10 s, irrespec-
tive of the configuration value.
The default value is 10 s.
The SIP registrar server configuration provides the following parameters:
• Registrar address
This is the IPv4 address or the fully qualified domain name (FQDN) of the
SIP registrar server. A newly configured address causes all ongoing calls
to be cleared.
There is no default value.
• Registrar port number
This parameter defines the SIP protocol UDP port number of the SIP reg-
istrar server. A newly configured port number is applied for new calls.
Currently active calls proceed using the previous port number.
• Registration mode
There exist several registration modes with a non gateway aware regis-
trar (a gateway aware registrar needs no registration since all subscribers
are already provisioned).

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IPSS4
IPSS4 supports the “one-by-one registration” mode where all subscriber

ports of the GW register individually one by one, and the “no registration”
mode where all subscribers have to be configured on the registrar by the
service provider.
A change from “one-by-one registration” to “no registration” causes no
further REGISTER messages after expiry of the current registration.
A change from “no registration” to “one-by-one registration” causes a
REGISTER message to be sent for each configured subscriber port.
The default value is one-by-one registration.
• Registration expiration time
When a client sends a REGISTER request, he suggests an expiration
interval to the registrar server that indicates how long the client would like
the registration to be valid. If the registrar does not accept the proposed
registration expiration time the registrar makes a proposal by its own
which will then be accepted by the IPSS4 gateway.
The registration is renewed with the first call after 90% of the registration
expiration time.
The default value is 100 hours.
Please note:
If a proxy server is provisioned then requests to the registrar are always sent
via the proxy server, i.e. the destination address on the IP layer is the proxy
address in this case. The proxy server will then forward the corresponding
messages to the registrar server.
Please note:
The registration process is controlled by some parameters of the custom
parameter set (CPS):
With the default values the gateway registers 50 subscribers in an interval of
5 seconds. The registration of 912 subscribers takes therefore 912/50*5 =
95 s.
If the registration fails a new registration attempt is started after 60 s.
Please note:
If an INVITE requests gets a 403 (Forbidden) response the IPSS4 can initi-
ate a re-registration for the particular subscriber.
This is controlled by the CPS parameters SIP1.24.
Please note:
The registrar address configuration is required even when the registration
mode is set to “No Registration”.
Please note:
The registration status is available on the unit layer (AP: /unit-x, Status -
Registration) for all unregistered subscriber numbers and on the port layer
(AP: /unit-x/portgroup-y/port-z, Status - Registration) for the PSTN, ISDN-BA
and hunting group ports.
5.5.2 Locating SIP servers
A preferred and an alternate DNS server IP address can be configured in the

Gateway - Network Interface configuration of IPSS4. With a DNS server con-

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IPSS4
figured IPSS4 provides the “locating SIP server” features according to

RFC3263.
Per default the IPSS4 unit accesses the preferred DNS server. In case of a
preferred DNS server access failure the IPSS4 unit switches to the alternate
DNS server. There is a DNS priority expiry timer provisioned via CPS. This
timer is started on DNS server switching. On timer expiry, IPSS4 reverts to
the preferred DNS server. The switching is non-revertive if the timer is set to
“0”.
When no proxy address is configured and a SIP request is pending, IPSS4
will send out an SRV request to the DNS server. The DNS server will then
send back a list of valid proxies where IPSS4 can select one of the proxies
to send the initial SIP request to.
Depending on the network scenario the following proxy, registrar and DNS
addresses have to be configured:
Table 20: SIP server address configuration
Network scenario Proxy Registrar DNS server
address type address type address type
No DNS server available IP IP or FQDN -
No proxy server available - IP or FQDN IP
DNS and proxy servers availa- IP or FQDN IP or FQDN IP
ble
The gateway behaviour for the above network scenarios is controlled by the
− configuration of the proxy and registrar addresses and port numbers,
and the
− configuration of the DNS server address.
5.5.2.1 Scenario without DNS server
When no DNS server is available the following configurations are required:

Table 21: DNS server configuration
Preferred DNS Alternate DNS server
server address address
empty empty
Table 22: Primary proxy configuration

Proxy address Proxy port number Behaviour
valid IP address port number ≠ 0 SIP requests are sent to the proxy
Table 23: Secondary proxy configuration

Table 24: Registrar configuration

Registrar address Registrar port num- Behaviour
ber
valid IP address port number ≠ 0 REGISTER requests are sent to
the proxy including the port num-
ber.

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Table 24: Registrar configuration (continued)

ber
valid IP address port number = 0 REGISTER requests are sent to
the proxy excluding the port num-
ber.
valid FQDN port number ≠ 0 Proxy performs an A record
lookup to get the registrar IP
address.
REGISTER requests are sent to
ber.
valid FQDN port number = 0 Proxy performs an SRV record
lookup to get a registrar IP
address and port number.
ber.
5.5.2.2 Scenario with DNS server and with proxy server
When both the proxy and the DNS servers are available the following config-
urations are required:
valid IP address valid IP address or
empty

valid FQDN port number ≠ 0 IPSS4 performs an A record
lookup to get the proxy IP address

lookup to get the proxy IP address

ber
ber.
ber.

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IPSS4

ber
valid FQDN port number ≠ 0 Proxy performs an A record
address.
ber.
valid FQDN port number = 0 Proxy performs an SRV record
If the port number ≠ 0 is config-
ured in the SRV record, the REG-
ISTER requests are sent to the
proxy excluding the port number.
If the port number = 0 is config-
ured in the SRV record, the DNS
lookup fails and no REGISTER
message is sent.
5.5.2.3 Scenario with DNS server and without proxy server address
When no proxy server address is available the following configurations are

required:
valid IP address valid IP address or
empty

empty port number = 0 IPSS4 performs an SRV record
lookup to get a proxy IP address
and port number

empty port number = 0 IPSS4 performs an SRV record
lookup to get a proxy IP address
and port number

ber
the registrar including the port
number.
the registrar excluding the port
number.

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IPSS4

ber
address.
the registrar including the port
number.
valid FQDN port number = 0 IPSS4 performs an SRV record
If the port number ≠ 0 is config-
ured in the SRV record, the REG-
ISTER requests are sent to the
proxy excluding the port number.
If the port number = 0 is config-
ured in the SRV record, the DNS
lookup fails and no REGISTER
message is sent.
Please note:
The proxy availability check shall still be executed even when no proxy
address is configured.
→ The OPTIONS or REGISTER message is sent to the registrar
address.
5.5.2.4 Scenario with different home domains
This scenario provides two different domains, e.g. one serving residential
customers and another serving business customers. Each domain has its
own proxies and registrars, i.e.
• Business home domain ‘businessprovider.com’ with SIP port number
5060:
− with proxy ‘proxy1B.businessprovider.com’, and
− with proxy ‘proxy2B.businessprovider.com’ (if required), and
− with registrar ‘businessprovider.com.
• Residential home domain ‘residentialprovider.com’ with SIP port number
5070:
− with proxy ‘proxy1R.residentialprovider.com’, and
− with proxy ‘proxy2R.residentialprovider.com’ (if required), and
− with registrar ‘residentialprovider.com.
The PSTN/ISDN-BA ports of one domain shall use the appropriate “Proxy /
Registrar Profile” and the “SIP Profile”.
The PSTN/ISDN-BA ports of the other domain have no profile selected (i.e.
set to “none”), meaning that the unit configuration is applied.
5.5.2.5 Scenario with different area codes
This scenario has the MileGate deployed at the area border of a network,
serving customers from two different areas with different area codes (e.g.
‘031’ and ‘032’).

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IPSS4
The PSTN/ISDN-BA ports assigned to the area code ‘031’ shall use the
appropriate “SIP Profile” with the area code configured to “31”.
The PSTN/ISDN-BA ports assigned to the area code ‘032’ have no profile
selected (i.e. set to “none”), meaning that the unit configuration is applied
with the area code configured to “32”.
Please note:
The SIP profile must use another SIP port number than the unit configura-
tion.
5.5.2.6 Scenario with manual A record and SRV record entries
In cases where the configured DNS server is not able to deliver the correct
address and service information or the DNS server is not available at all, the
required address (A) records and service (SRV) records can be configured in
the IPSS4 unit configuration dialogue.
The network scenarios described above define if the required records are A
records or SRV records. In case a SRV record is required also the corre-
sponding A records must be added to provide the IP addresses to the IPSS4
unit.
An A record requires the configuration of the
• Domain name for which this record is valid:
E.g. proxy-a.london.voip.com.
• IP address:
E.g. 172.31.63.181.
An SRV record requires the configuration of the
• Symbolic name of the service:
Fixed to “SIP”.
• Transport protocol of the service:
Fixed to “UDP”.
• Domain name for which this record is valid:
E.g. london.voip.com. The default name is empty.
• Priority of the target host:
Valid range 0 … 65’535. Lower values mean higher priority.
• Relative weight for records with the same priority:
Valid range 0 … 65’535. Higher values mean higher probability of being
selected.
• The UDP port on which the service is to be found:
The valid port range is 0 … 65’535.
• Domain name of the target host:
e.g. proxy-a.london.voip.com. The default target name is empty.
Please note:
It is typically not required to manually add DNS records in a proper network
setup.
→ You can leave the A and SRV records tables in the DNS table configu-
ration empty.
Please note:
Manually adding or removing DNS records flushes the DNS cache.

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IPSS4
Please note:
Manually added DNS records are applied even when the local resolver
cache is disabled by the CPS.
5.5.3 Penalty box
IPSS4 implements a penalty box algorithm, i.e. destination addresses (proxy

or registrar) that do not respond are added to a penalty box together with
their port number. This ensures that unavailable destination addresses will
not be considered for a certain time. IPSS4 then selects the next destination
address based on the priority and weight and tries sending the message
again.
The time a destination address remains in the penalty box is provisioned via
the CPS in the range from 1 to 600 s. It is also possible to disable the pen-
alty box algorithm completely via CPS.
The penalty box status is available in the AP: /unit-x/status - gateway provid-
ing the IP address, the port and the time to live of the penalty box entries.
In case all destination addresses for a certain proxy or registrar are in the
penalty box then the last used destination address is used to send the SIP
message to. This ensures that the proxy availability check is always exe-
cuted. Other destination addresses than the last one are only accessed
when they have left the penalty box.
Please note:
The penalty box algorithm is applied to resolved targets from a DNS query
and also to provisioned IP addresses.

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IPSS4
5.6 Direct dialling in and trunk hunting
IPSS4 supports trunk hunting. Trunk hunting enables B-channels from one
to 16 ISDN-BA user ports to be grouped in a hunting group. An incoming call
for a trunk hunting group number will be terminated on any free B-channel of
one of the hunting group members.
The benefit of trunk hunting is the effective connection of ISDN-PABXs to the
gateway reducing the number of required subscriber lines.
All members of the hunting group use the same set of profiles.
The hunting group can be allocated
• one directory number or
• a range of up to 100 directory numbers using the “DDI Numbers” hunting
group port type, supporting the registration of all provisioned numbers, or
• up to 1000 directory numbers using the “DDI Ranges” hunting group port
type, supporting the registration of the default number only.
A trunk hunting group can be used with the direct dialling in (DDI) service.
DDI enables a caller on the public network to dial in directly to an extension
of an ISDN enabled PABX without the assistance of a receptionist. If the
number dialled is busy or does not answer, PABX functions can be used to
connect the caller to the receptionist, a secretary or a voice messaging sys-
tem.
One of the DDI numbers is designed as the subscribers “default ISDN num-
ber”. This number is registered in the registrar for identification purposes. On
IPSS4 when using the “DDI Numbers” hunting group port type the first num-
ber in the list of DDI numbers becomes the default number. With the “DDI
Ranges” hunting group port type the default number is configurable. The
default number is either registered as global i.e. including country and area
code or as local i.e. without country and area code.
Please note:
With the “DDI Numbers” hunting group port type, registration is configurable
to be only applied for the default number or for all subscriber numbers.
Please note:
With the “DDI Ranges” hunting group port type, registration is fixed to be
only applied for the default number.
If only the default number is registered IPSS4 applies the “authorisation user
name” and the “authorisation password” of the default number for the HTTP
digest authentication.
Incoming direction:
As overlength numbers are handled for huntgroups, i.e. numbers comprising
more digits than configured as subscriber number, one can provision the DDI
prefix as only number. All numbers with the same prefix will then be routed
to the respective huntgroup.
The “DDI Prefix Length” parameter in the hunting group configuration causes
the truncation of the DDI prefix. Only the DDI part of an incoming call num-
ber will be sent to a PABX connected to the hunting group interface.
As a first example assume the hunting group DDI numbers are in the range
+41 31 900 1200 to +41 31 900 1299, and the DDI prefix length is configured

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IPSS4
to be 3. From an incoming call for the number +41 31 900 1225, IPSS4 will
truncate
− the country code +41,
− the area code 31, and
− 3 digits of the remaining number, i.e. 900.
The short number sent to the PABX is therefore 1225.
As a second example assume the hunting group has one DDI number
+41 31 900 1200 and the connected PABX uses a two digit extension to
access the connected subscribers. The DDI prefix length is configured to be
7. From an incoming call for the number +41 31 900 1200-34, IPSS4 will
truncate
− the country code +41,
− the area code 31, and
− 7 digits of the remaining number, i.e. 900 1200.
The short number sent to the PABX is therefore the extension number 34.

It is the responsibility of the network provider to configure a “DDI Prefix
Length” which generates uniquely identifiable short numbers from the DDI
numbers list.
Outgoing direction:
If the received calling party number from the PABX matches the last digits of
one of the configured numbers, this configured number is sent out as calling
party number.
When no match is found the IPSS4 unit offers two possibilities for the calling
party number insertion, controlled by the CPS:
• The default number is sent out as calling party number.
• The received calling party number from the PABX is prepended with the
configured DDI prefix number.

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IPSS4
5.7 Equipment protection (EQP)
5.7.1 EQP features
To protect the SIP gateway against a failure on the gateway unit IPSS4,
MileGate offers the possibility to backup a (working) IPSS4 unit with a sec-
ond (protecting) IPSS4 unit (1:1 equipment protection). In case of a failure
on the active (working or protecting) unit the user traffic is rerouted from the
failed IPSS4 unit to the stand-by IPSS4 (protecting or working) unit.
A status dialogue allows the operator to control and monitor the equipment
protection.
These functions are controlled on the unit level and require no special con-
figuration on the NE level.
Please note:
The protection switching is non revertive, i.e. after the repair of a failed
IPSS4 unit, the currently active IPSS4 unit remains the active unit irrespec-
tive if it is the working or protecting unit.
Please note:
During a protection switching event the user traffic is interrupted for up to
85 s.
5.7.2 EQP prerequisites
To enable equipment protection for the IPSS4 unit some prerequisites must
be met:
• The protecting IPSS4 unit must be in the unassigned state. Otherwise the
unit will not be selectable in the EQP configuration in AP: /unit-x, Configu-
ration - EQP: Create EQP Group…, EQP Group Creation, Protecting Unit
of the working unit.
• The protecting unit must be hardware compatible with the working unit.
Check the hardware compatibility status after the EQP group configura-
tion in the AP: /unit-x, Status - EQP: Units Status, HW Compatible.
The following requirements must be fulfilled that the two units are stated
as hardware compatible:
− Identical unit function. The unit function is composed of the 5 first
characters of the unit name, e.g. IPSX4. The unit name is available at
the AP:/ Main - Equipment, Unit.
− Identical board identification, e.g. 308. The board identification is avail-
able at the AP:/ Main - Inventory, Board ID.
− Identical hardware variant. The hardware variant is the truncation of
the hardware key divided by 256, e.g. 2/256 = 0. The hardware key is
available at the AP:/ Main - Inventory, Hardware Key.
• The protecting unit must be software compatible with the working unit.
Check the software compatibility status after the EQP group configuration
in the AP: /unit-x, Status - EQP: Units Status, SW Compatible.
The following requirements must be fulfilled that the two units are stated
as software compatible:

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IPSS4
− Identical unit function. The unit function is composed of the 5 first

characters of the software name, e.g. ipss4. The unit name is availa-
ble at the AP:/ Main - Equipment, Software.
In order to guarantee the full compatibility of all features it is strongly rec-
ommended to use the same software on the working and on the protect-
ing unit.
The compatible software must be installed on the protecting IPSS4 unit
before the EQP group creation.
• The unit configuration of an equipment protection group is always done
on the active unit. The configuration on the standby unit is not possible.

The configuration of an equipment protection group must be saved (ECST
NE menu or ECST tool bar: Save to NE).
→ Otherwise, in case of a protection switch event, the NE will load an
outdated configuration from the internal configuration database and
the NE configuration will be corrupted.
5.7.3 EQP configuration
The working IPSS4 unit of an EQP group is assigned and configured the
same way as a stand alone IPSS4 unit.
The protecting IPSS4 unit is running with the same ESW as the working unit
and must be in the unassigned state.
The 1+1 equipment protection group is configured on the working unit:
• AP: /unit-x, Configuration - EQP.
− Execute the command “Create EQP Group…”.
− Select the Protecting unit, e.g. /unit-21.
− Execute “OK”.
• Save the NE configuration.
Further on any changes on the IPSS4 configuration must be done on the
active unit. To find out which unit is the active unit check the AP tree or the
unit status of the working or protecting IPSS4 unit.
5.7.4 EQP status
The unit status of the working and protecting units shows the actual status of
the units belonging to the equipment protection group. The unit status offers
also the commands for the EQP manipulation:
• Manual switch
The currently standby unit is set as active unit and the currently active
unit is set as standby unit. This requires that the currently standby unit is
in operational state, i.e.
− has no failure,
− is not isolated.
A manual switch is possible if it is indicated with the “Manual Switch-Over
Allowed” parameter.
Note that this command can only be activated on the working unit.

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IPSS4
• Forced switch
The currently standby unit is set as active unit, independent of the opera-
tional state of the currently standby unit.
Note that there is a risk that the user traffic will be permanently inter-
rupted if the currently standby unit is not operational.
The currently active unit is set as standby unit.
Note that this command can only be activated on the working unit status
window.
• Isolate unit
To be able to perform a maintenance action, e.g. update of the embed-
ded software, on an active unit without activating a protection switch, the
working unit can be isolated. This means that the protection switching
state machine is frozen and no protection switch will be done until the iso-
lation of the unit is removed.
Note that the isolate unit command can only be applied to the working
unit.
• Join unit
Remove the isolation of a previously isolated unit.
Note that the join unit command can only be applied to the working unit.
The table in the EQP status window displays the following items:
• Unit
MO address of the unit belonging to the EQP group.
• EQP unit mode
The working unit is the unit where the protection group has been config-
ured.
The protecting unit is the unit that has been set to the unassigned state
before configuring the protection group.
• Active
Active true means the unit is the active unit, i.e. it is the operational unit.
Active false means the unit is the standby unit, i.e. it is not the operational
unit.
The active state can be changed with the “Manual Switch” and “Forced
Switch” commands.
• Failure
Failure true means the unit is in a failure state.
Failure false means the unit is not in a failure state.
The failure state can not be changed manually.
• Substituted
Substituted true on the working unit means the unit has been substituted
by the protecting unit. A substituted unit is also in the “active false” state.
Substituted false on the working unit means the unit has not been substi-
tuted, i.e. it is the active unit or it has been isolated.
The substituted state of the protecting unit is always false.
• Isolated
Isolated true means the unit has been isolated with the “Isolate Unit”
command.
Isolated false means the unit is not isolated.
The isolation state can be changed with the “Isolate Unit” and “Join Unit”
commands.
The isolated state of the protecting unit is always false.
• HW Compatible

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IPSS4
HW compatible true means the working HW unit is compatible with the

protecting HW unit.
HW compatible false means the working HW unit is not compatible with
the protecting HW unit. Equipment protection is not possible.
• SW Compatible
SW compatible true means the working unit embedded software (ESW) is
compatible with the protecting unit ESW.
SW compatible false means the working unit ESW is not compatible with
the protecting unit ESW. Equipment protection is not possible.
• DB Saved
DB saved true means the current configuration of the working unit has
been saved to the MileGate internal database.
DB saved false means the current configuration of the working unit has
not been saved to the MileGate internal database. A protection switching
event will load an outdated configuration and traffic will be disturbed.

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IPSS4
5.8 Synchronization
All voice applications in a network require the synchronization to one com-

mon clock source, the PSTN network timing, which is a primary reference
clock (PRC).
If (the TDM part of) an NE is not synchronized to the network synchroniza-
tion source, the MileGate will run on its internal clock source. Byte and frame
slips will occur periodically as a consequence. These disturbances do not
affect the transmission of digitized voice to a great amount, but a frame slip
can lead to loss of data or invalid data during data transmission.
Each NE requires therefore its plesiochronous equipment timing source
(PETS) as an NE synchronization source, where the PETS itself is synchro-
nized to the network synchronization source.
For more information about the synchronization of a MileGate NE please
refer to [314] User Guide “TDM Services and Cross Connections”.
5.9 Ethernet switch
The IPSS4 unit accesses the MileGate chassis via its backplane 1 GbE
ports. The internal port of the IPSS4 unit is available as external port of the
MileGate chassis switch. The port reference is the corresponding internal
port of the COGE5 unit:
• AP: /unit-11 (COGE5)/iports/iport-y
• AP: /unit-13 (COGE5)/iports/iport-y
The IPSS4 MileGate chassis switch port must be configured to a port mode
supporting tagged traffic. Please refer to section 5.4.6.3 "VLAN for the net-
work interface" (on page 72) and refer to [341] Quick Guide “Ethernet
Switching”.

The IPSS4 MileGate chassis switch ports to the active COGE5 and to the
redundant COGE5 must be configured identically.
→ Otherwise the switch behaviour will change after a COGE5 switcho-
ver.
Please note:
The control connection and the media connection use the same VLAN and
the same MileGate chassis switch port mode.

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Commissioning User Manual
IPSS4
6 Commissioning
This section describes the management of the basic system functions of the
IPSS4 unit and the configuration of the media and control links:
• Section 6.1 "Customer profiles" (on page 93) describes the custom
parameter set (CPS) handling.
• Purpose and commissioning example of the profiles used for the IPSS4
unit (section 6.2 "Profiles" (on page 95)).
• Commissioning example of the IPSS4 unit and the configuration of the
SIP gateway with one PSTN unit and one ISDN-BA unit (section 6.3
"Commissioning of a gateway" (on page 98)).
• Commissioning example of a SIP gateway with equipment protection
(section 6.4 "Gateway configuration with equipment protection" (on
page 113)).
Please refer to [355] User Manual “ECST” for details on the general GUI
aspects, and to [302] User Guide “MileGate 2310/2510” for specific charac-
teristics of the MileGate.
6.1 Customer profiles
The IPSS4 unit configuration includes a set of pre-provisioned parameters,

the so called Custom Parameter Set (CPS). The CPS defines the PSTN
national mapping, i.e. parameters for the SIP and RTP protocol implementa-
tion as well as traffic related parameters (codec, tone, ringing signal parame-
ters etc.). Most of these parameters must be set to the same value for all
access gateways in the same network.
KEYMILE will provide the CPS file to customers. A questionnaire is used to
capture the appropriate information.
KEYMILE has designed the CPS to provide an easy way of handling bundles
of parameters with the same value for all IPSS4 units in a certain network
area. The network area with the same CPS could be a country, a network
operator’s business domain or a network with the same softswitch type.
Once the CPS is defined, the whole parameter bundle is available for each
IPSS4 unit in the network area and can be selected with a keystroke.
The CPS for the IPSS4 unit has the prefix “IPSS_”.
Please note:
The application of a CPS with the modification of certain parameters, marked
with a “#”-prefix in the parameter description, requires a restart of the IPSS4
unit.
→ For further information refer to [808] Technical Bulletin “MileGate CPS
Handling Procedures”.
For the handling of the CPS with the ECST CPS tool please refer to [355]
User Manual “ECST”.

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IPSS4
6.1.1 Custom parameter set (CPS) structure
The CPS includes more than 100 parameters which are structured in the fol-
lowing groups:
Table 33: CPS parameter groups
Group abbreviation Group name Group description and examples
VP and VPC Voice processing Voice Processing parameters, e.g. gain values and control of sup-
ported features, e.g. comfort noise generation on/off
IP IP Internet Protocol parameters, e.g. TTL value
RTC RTCP (Real Time Con- Parameters for RTCP enabling and disabling
trol Protocol)
HS Hook switch PSTN Hook Switch detection timing values
PGP PSTN general parame- General PSTN signalling parameters
ters
CR Cadence ringing PSTN cadence ringing sequence definitions
PS Pulsed signal definition PSTN pulsed signal timing values
PDM Dial pulse to digit map- Mapping of dial pulse numbers to digits
ping
SIP SIP protocol Process parameters, e.g. timer values
PRCT PSTN release cause Tone configuration for different PSTN release causes
tones
DNS DNS DNS specific parameters
SRV Supplementary ser- Supplementary services and digit map specific parameters
vices
DEV Deviation Deviations from standards
6.1.2 Defining the CPS
The content of the CPS depends on national or network operators require-

ments and is related to the SIP implementation of the softswitch type. There-
fore it makes sense that the network operator defines the content of the CPS
since he has the knowledge about all requirements.
The document [913] Technical Bulletin “SIP / IPSS4 questionnaire MileGate”
contains a list of all available parameters together with their value ranges
and it is designed to define the CPS content.
You may order the questionnaire at your nearest KEYMILE sales office.
After filling in the questionnaire send it to the contact address contained in
the document. KEYMILE will then create the corresponding CPS file.
Please note:
You must not change the CPS file with an xml editor or by any other means.
→ ECST authenticates the selected CPS file prior to loading and refuses
any altered or invalid file.

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IPSS4
6.2 Profiles
6.2.1 General
A profile is a set of configuration parameters that can be applied to a certain

managed object.
Profiles are handled offline with the Profile Tool contained in the ECST
(Menu: Tools / CPS & Profile …). The profile tool allows the operator to
− create new profiles from profile templates or from existing profiles and
store them in the element manager,
− download profiles (from the element manager to the MileGate) and
upload profiles (from the MileGate to the element manager),
− import profiles from the file system and export profiles to the file sys-
tem,
− view profiles, and
− delete profiles on the element manager and on the network element.
Please note:
The creation of profiles is based upon templates that are provided with the
ESW versions. The templates are available only after importing the respec-
tive service unit’s ESW in ECST.
Please note:
Make sure that you only use profiles from templates that have been installed
with the ESW running on the respective unit. If you use profiles from other
ESW versions, you may get an error message when trying to apply the pro-
file to the unit’s configuration.
The availability and valid range of all profile configuration parameters is
listed in the reference section of this document. Please refer to section 8.2
"Profiles" (on page 131).

Applying a new profile will lead to a service interruption.
For the handling of the profile files with the ECST please refer to [355] User
Manual “ECST”.
6.2.2 IPSS4 profiles
The IPSS4 unit supports 6 different profile types:

• SIP profile
SIP gateway configuration, e.g SIP port number, country code, etc.
• Proxy/Registrar profile
Primary and secondary proxy and registrar configuration.
• Codec/SDP profile
Voice, SDP and codec configuration.
• PSTN port profile
Supplementary services configuration.

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IPSS4
• Enterprise profile
Enterprise supplementary services configuration.
• ISDN-BA port profile
Supplementary services configuration.
Please note:
ISDN-BA subscriber units will be available in a future release.
All parameters defined in one of the above profiles are selected by configur-
ing the corresponding profile name in the port or group configuration dia-
logue.
Please note:
IPSS4 supports the provisioning of up to 8 different profiles for the SIP pro-
file type. The number of profiles for the other profile types is not restricted.
The profiles are applicable to the managed objects according to the following
table.
Table 34: Profile applicability to managed objects
Profile Profile type Applicable to MO
SIP SipProfile_1.00.00 unit-x/portgroup-y/port-z (PSTN)
unit-x/portgroup-y/port-z (ISDN-BA)
unit-x/huntgroups/group-a (ISDN-BA)
Proxy/Registrar SipProxyRegistrarProfile_1.04.00 unit-x/portgroup-y/port-z (PSTN)
Codec/SDP SipCodecSdpProfile_1.01.00 unit-x/portgroup-y/port-z (PSTN)
PSTN port SipPstnProfile_2.02.02 unit-x/portgroup-y/port-z (PSTN)
Enterprise SipEnterpriseProfile_1.00.00 unit-x/portgroup-y/port-z (PSTN)
ISDN-BA SipIsdnBaProfile_2.01.00 unit-x/portgroup-y/port-z (ISDN-BA)
In the case it is not required to assign individual parameter values to the

PSTN or ISDN-BA ports or to the huntgroup groups, but instead to use the
same parameter values for all ports or groups, the profile selection in the
port or group configuration dialogue can be set to “none”. In this case the
parameters configured on the IPSS4 unit layer are used instead of the profile
defined parameter.

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IPSS4
<ap >
MileGate parameter a of port-z
is defined by the unit
configuration
0...2 <ap > <mf>
unit-x: IPSX4 Configuration
<ap>
<attribute>
portgroup-y:
parameter a
PSTN
<ap>
<mf>
port-z Configuration
parameter a
<attribute>
Profile „none“
<ap>
<profile> <mf>
port-zz
prof_1 Configuration
parameter a
<attribute> <attribute>
parameter a Profile „prof_1“
parameter a of port-zz
is defined by the profile
prof_1
Figure 28: Parameter assignment from profiles and from the unit configura-
tion

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IPSS4
6.3 Commissioning of a gateway
6.3.1 User port to subscriber number reference
In the SIP protocol, the subscriber numbers are configured directly in the SIP
gateway.
In the MileGate the user ports are located on the linecard units. The connec-
tions from the user ports to the IPSS4 unit have to be configured during the
commissioning of the gateway.
The subscriber number is assigned to the user port access point (AP) on the
IPSS4 unit. The subscriber number of a user port on a linecard is then
defined by the connection between the user port on the linecard and the user
port on the IPSS4 unit.
The example below shows that the port-1 of the SUPM1 unit has the sub-
scriber number “1001”, as it is configured on the IPSS4 unit.
The example below shows that the port-1 of the SUIx1 unit has the sub-
scriber number “2001”, as it is configured on the IPSS4 unit.
AP: /unit-19/portgroup-1/port-1,
Configuration:
Cross Connection 1 Subscriber Number: 1001
MileGate
Ethernet star
SUPM1 IPSS4 COGE5

port-1 TDM to packet Bridging GbE IP
conversion Network
port-2
port-16
AP: /unit-19/portgroup-2/port-1,
Configuration:
SUIx1 Subscriber Number: 2001
port-1
port-2
PBUS
Cross Connection 17
port-16
Figure 29: Subscriber number to port assignment in MileGate
Please note:
It is recommended to give the IPSS4 portgroups the same index number as
the linecard slot number, e.g. give the portgroup the index number 1 which
will be connected to a linecard plugged in slot 1 of the MileGate subrack.
→ This facilitates the check of the cross connections between the line-
card and the IPSS4 unit.

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IPSS4
Please note:
It is recommended to create the IPSS4 portgroups with the same number of
ports as has the linecard it will be connected to, e.g. create a portgroup with
16 ports which will be connected to a SUPM1 unit.
→ This facilitates the set-up and check of the cross connections between
the linecard and the IPSS4 unit.
Please note:
It is recommended to create the cross connections between the ports on the
linecard and the ports in the IPSS4 portgroup with the same port index num-
ber, e.g. connect port-1 on the linecard to port-1 of the portgroup.
→ This facilitates the check of the cross connections between the line-
card and the IPSS4 unit.
For the configuration of cross connections in the MileGate please refer to
[314] User Guide “TDM Services and Cross Connections”.
6.3.2 Set-up for the implementation of the NGN voice service
Figure 30 shows the set-up for the implementation of the NGN voice service
for several PSTN and ISDN-BA subscribers in the SIP application. The sub-
scribers analogue telephone sets are all connected to a SUPM1 unit. The
subscribers ISDN telephone sets are all connected to a SUIx1 unit.
The user ports are cross connected MileGate internally to the gateway unit
IPSS4. The IPSS4 Ethernet traffic is then connected to the MileGate chassis
switch via the GbE star:
• The IPSS4 network interface uses one VLAN which is connected to the
internal port of the COGE5 unit.
• Media and control traffic use the same VLAN.
The SFP-based optical GbE interface 1 on the COGE5 unit is used for the
uplink towards the IP network.

Voice services will not work reliably if the link speed to the IP network is not
at least 100 Mbit/s and full duplex.

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IPSS4
Registrar
172.31.32.121/24
Access Gateway
“xxx” myprovider.com
Proxy
172.31.32.121/24
IP
Network
172.31.39.66/29
MileGate 2510
Access Gateway
S C S I
U O U P “Rome”
P G I S 172.31.39.68/29
M E x S
1 5 1 4
# 1001 Port 3 Port 2 # 2001
# 1002 Port 8 Port 5 # 2002
# xxxx = subscriber call number
Figure 30: Set-up for the implementation of the NGN voice service
Table 35: PSTN and ISDN-BA port mapping for the access gateway Rome
Linecard Portgroup
Slot number Port number Index Port number Subscriber number
5 port-3 5 port-3 1001
5 port-8 5 port-8 1002
12 port-2 12 port-2 2001
12 port-5 12 port-5 2002
Please note:
The subscriber call numbers have to be configured in the MileGate configu-
ration.
Table 36: Registrar configuration for the access gateway Rome

Access gateway Device URI registration Password
Rome sip:+3961001@172.31.39.68 1234
Rome sip:+3961002@172.31.39.68 1234
Rome sip:+3962001@172.31.39.68 1234
Rome sip:+3962002@172.31.39.68 1234

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The above table assumes that the country code is +39 and the area code is
6 in the gateway and that the registration is done as global. The registration
in the registrar is done with the device URI including the country and area
code. The password “1234” is the same for all subscribers.
6.3.3 Prerequisites
Before starting the commissioning of a SIP gateway on the IPSS4 unit as

shown in Figure 30, the following prerequisites need to be fulfilled.
• A COGE5 needs to be in operation in slot 11 of the MileGate subrack.
• A PSTN linecard SUPM1 needs to be in operation in the MileGate sub-
rack.
• An ISDN-BA linecard SUIx1 needs to be in operation in the MileGate sub-
rack.
• The IPSS4 service unit is inserted into one of the following slots:
MileGate 25x0:
− slot 1 … 10,
− slot 12 … 21.
MileGate 23x0:
− slot 7 … 10,
− slot 12 … 14.
For the installation of the unit, refer to [301] User Guide “MileGate 2510
Installation” or refer to [310] User Guide “MileGate 2310 Installation”.
• Optionally one or several user port profiles can be created and down-
loaded to the MileGate. All parameters available in the profiles can also
be configured globally on unit level, i.e. these parameters are then valid
for all user ports not using profiles. In the following it is assumed that all
parameters are configured directly on the IPSS4 unit. No profiles are
required.
• You need a custom parameter set (CPS) appropriate for your application
and your PSTN network. The CPS must be ordered at your KEYMILE
representative. The CPS must be downloaded to your MileGate NE to be
selectable in the IPSS4 configuration.
• The element manager ECST needs to be installed on a PC, and a man-
agement connection from the ECST to the MileGate needs to be up and
running. For details about the installation and operation of the ECST
please refer to [354] Quick Guide “ECST”.
The amount and accessibility of operations depend on the user profile
with which you are logged in. For more information, please refer to [323]
User Guide “Management Communication”.
• The PETS on the MileGate network element must be configured to an
appropriate synchronization clock source, i.e. a clock source traceable to
a PRC. For the PETS configuration refer to [314] User Guide “TDM Ser-
vices and Cross Connections”.
6.3.4 Configuration of the gateway
For the configuration of the SIP gateway consisting of a single MileGate net-
work element, the following steps have to be performed. All network specific
parameters have to be adapted according to the application. The main steps
are:

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• Create the profiles (optional)

• Download the profiles (optional)
• Download the CPS
• Assign the unit
• Download and start the ESW
• Configure the gateway SIP part
• Configure the gateway supplementary services part
• Configure the gateway media path
• Configure the gateway local network interface
• Configure the gateway DNS table (optional)
• Configure the proxy and registrar parameters
• Configure the digit map parameters
• Select the CPS file
• Create a PSTN and an ISDN-BA port group
• Configure the PSTN and ISDN-BA port parameters
• Configure the PSTN and the ISDN-BA linecards
• Create the cross connections from the ports in the portgroups to the
PSTN and the ISDN-BA ports on the linecards
• Configure the internal chassis switch port
• Configure the Ethernet uplink port
• Configure the NE synchronization
• Save the configuration and restart the IPSS4 unit
Gateway configuration This action list shows step by step how to configure the gateway (GW). The
GW accesses 2 PSTN (SUPM1) and 2 ISDN-BA (SUIQ1) subscribers.
The following assumptions and identifiers are used:
- The IPSS4 unit is assumed to be plugged in slot 19 of the MileGate 2510.
- The PSTN linecard SUPM1 running with a proper ESW is plugged in slot
5 of the MileGate 2510.
- The ISDN-BA linecard SUIx1 running with a proper ESW is plugged in slot
12 of the MileGate 2510.
Create the profiles (optional) Proceed as follows:

1. Refer to section 6.2 "Profiles" (on page 95) for the profile creation.
2. Create one or more profiles for the SIP configuration parameters.
3. Create one or more profiles for the proxy and registrar configuration
parameters.
4. Create one or more profiles for the codec and SDP configuration param-
eters.
5. Create one or more profiles for the PSTN port configuration parameters.
6. Create one or more profiles for the enterprise configuration parameters.
7. Create one or more profiles for the ISDN-BA configuration parameters.
 All required profiles have been created.

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Download the profiles Proceed as follows:

(optional)
1. Open the CPS & Profile tool of the ECST (Menu: Tools / CPS &
Profile …):
2. Select the profiles file in the “Local Profiles” area of the CPS & Profile
tool by clicking on the file names while pressing the Ctrl-key.
 The file name rows are highlighted.
3. Press the download button:
 The files are downloaded and appear in the “Remote Profiles” area
of the CPS & Profile tool.
4. Execute “Close”.
 All required profiles have been downloaded from the element manager to
the network element.
Download the CPS Proceed as follows:

1. Open the CPS & Profile tool of the ECST (Menu: Tools / CPS &
Profile …):
2. Select the CPS file with a filename: IPSS_xxx in the “Local Profiles” area
of the CPS & Profile tool by clicking on the file name.
 The file name row is highlighted.
3. Press the download button:
 The file is downloaded and appears in the “Remote Profiles” area of
the CPS & Profile tool.
4. Execute “Close”.
 The CPS file is now available in the MileGate NE.
Assign the IPSS4 unit Proceed as follows:

1. Navigate to the equipment status dialogue:
 AP:/unit-19, Main - Equipment:
2. Assign the unit:
 Execute the command “Assign”.
Please note:
This command has to be executed only when the property “Assignment Sta-
tus / State” has the value “Unassigned”, i.e. after the first insertion of the
IPSS4 unit.
 The unit is assigned and can be configured now.

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Download and start the ESW Proceed as follows:

1. Open the Software Download tool of the ECST (Menu: Tools / Software
Download …):
2. In the table row of the IPSS4 unit click on the “Software to install” col-
umn.
3. Select the ESW from the drop down list.
4. In the table row of the IPSS4 unit tick the “Delete unused Software” box.
5. Press the “Download and Start …” button.
6. In the “Parameters for command Download and Start” select the algo-
rithm “Upgrade Units Only”.
7. Click “OK”.
 The ESW is downloaded and the IPSS4 unit starts with the config-
ured ESW.
 The IPSS4 unit is now running with the correct ESW.
Configure the Gateway SIP Proceed as follows:

part 1. Navigate to the unit configuration dialogue:
 AP:/unit-19, Configuration - Gateway - SIP.
2. Enter the SIP gateway parameters:
 Gateway Name = Rome.
 Home Domain = myprovider.com.
 SIP Port Number = 5060.
 Country Code = +39.
 Area Code = 6.
 Retransmission Timer T1 = 500 ms.
 Max Retransmission Interval T2 = 4 s.
 SIP Extension 100rel Required = false.
 Asserted Identity Mode = None.
 Overlap Signalling = false.
 Overlap T10 timer = 10 s.
 UAC Request Session-Timer = false.
 UAS Request Session-Timer = false.
 Session Expiration = 1800 s.
3. Execute “Apply”.
 The SIP gateway parameters are fully configured.
Configure the supplementary Proceed as follows:

services
1. Navigate to the configuration dialogue:
 AP:/unit-19, Configuration - Gateway - Supplementary Services -
Available Services.
2. Configure the supplementary services:
 Hotline = true.
 Per Call CLIR = true.
 Call Forwarding Unconditional = false.
 Call Forwarding Busy = false.
 Call Forwarding No Reply = false.
 Calling Line ID Presentation = true.
 CLIP Mode = FSK Transmission.
 Service Control = On Proxy.
 European Standard Enabled = true.

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 The available supplementary services are fully configured.
Configure the service codes Proceed as follows:

 AP:/unit-19, Configuration - Gateway - Supplementary Services -
Service codes.
2. Enter the service codes for the supplementary services:
 Toggle Or Accept Call = 2.
 Reject Call = 0.
 Disconnect Call = 1.
 Transfer Call = 4.
 3-Way Call = 3.
3. Configure the call waiting supplementary service:
 Activate = *43#.
 Deactivate = #43#.
 Initially Activated = false.
4. Configure the hotline supplementary service:
 Activate = *53*.
 Termination Character = #.
 Interrogate = *#53#.
 Start Timer = 5 s.
5. Configure the per call CLIR supplementary service:
6. Configure the call forwarding unconditional supplementary service:
7. Configure the call forwarding busy supplementary service:
8. Configure the call forwarding no reply supplementary service:
 The service codes for the supplementary services are fully configured.
Configure the media path, Proceed as follows:

part 1 1. Navigate to the configuration dialogue:
 AP:/unit-19, Configuration - Gateway - Media - Codec / SDP.
2. Configure the voice parameters:
 G.711 Silence Suppression = false.
 G.729AB Silence Suppression = true.

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3. Configure the SDP advanced parameters:

 T.38 Support = true.
 DTMF Relay Support = true.
 Telephone-Event Payload Type = 96.
 Clearmode Payload Type = 97.
 DTMF Relay Type = RFC 2833.
4. Select the preferred voice codec:
 Voice Codec 1 = G.711 A.
 The codec and SDP part of the media path are configured.
Configure the media path, Proceed as follows:

part 2
 AP:/unit-19, Configuration - Gateway - Media - Jitter Buffer / RTP.
2. Configure the adaptive jitter buffer parameters:
 Nominal Threshold = 50 ms.
 Fax+Clearmode Threshold = 80 ms.
 Low Threshold = 0 ms.
 High Threshold = 200 ms.
 Response Time = 2.5 s.
3. Configure the RTP port number range:
 Start Port Number = 50000.
 Stop Port Number = 54000.
 The jitter buffer and RTP part of the media path are configured.
Configure the local network Proceed as follows:

interface 1. Navigate to the configuration dialogue:
 AP:/unit-19, Configuration - Gateway - Network Interface.
2. Configure the IP parameters of the gateway:
 Gateway IP Address = 172.31.39.68.
 Subnet Mask = 255.255.255.248.
 Default Gateway = 172.31.39.10.
3. Configure the DNS server parameters:
 Preferred DNS Server = 172.31.63.181.
 Alternate DNS Server = 172.31.63.182.
4. Configure the QoS parameters:
 DSCP Control SIP = 40.
 DSCP Voice / VBD / RFC2833 = 46.
 DSCP Clearmode = 40.
 DSCP T.38 = 40.
 The network interface, DNS server and DiffServ parameters are fully
configured.
Configure the network inter- Proceed as follows:

face VLAN 1. Navigate to the configuration dialogue:
 AP:/unit-19, Configuration - Gateway - VLAN.

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2. Configure the VLAN parameters of the network interface:

 ID = 101.
 Priority = CoS 6 (Voice).
 The network interface VLAN parameters are fully configured.
Configure the DNS table Proceed as follows:

(optional)
 AP:/unit-19, Configuration - Gateway - DNS Table.
Please note:
The DNS table configuration is only required when the DNS server is not
able to resolve the FQDN of the proxy and registrar.
Configure the proxy and reg- Proceed as follows:

istrar association 1. Navigate to the configuration dialogue:
 AP:/unit-19, Configuration - Proxy/Registrar.
2. Configure the primary proxy:
 Proxy Mode = Primary Only.
 Proxy Address = 172.31.32.121.
 Proxy Port = 5060.
3. Configure the secondary proxy:
 Proxy Address = <empty>.
 Proxy Port = 0.
4. Configure the proxy availability check parameters:
 Proxy Availability Check, Enabled = true.
 Proxy Availability Check, Method = OPTIONS.
 Proxy Availability Check, Interval = 10 s.
5. Configure the registrar:
 Registrar Address = 172.31.32.121.
 Registrar Port = 5060.
 Registration Mode = One-By-One Registration.
 Registration Expiration Time = 100 h.
 The proxy and registrar parameters are fully configured.
Configure the digit maps Proceed as follows:

 AP:/unit-19, Configuration - Digit Map.
2. Add the first digit map entry:
 Click the “Add...” button.
 URI Schema = sip.
 Digit Map = xxxx.
 Domain / Phone Context = myprovider.com.
 Pre Strip = 0 digits.
 Pre Pend = <empty>.
 Execute “Apply”.

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3. Add the second digit map entry:

 Digit Map = 6xxxx.
 Domain / Phone Context =myprovider.com.
 Execute “Apply”.
4. Add the third digit map entry:
 Digit Map = x. .
 Domain / Phone Context = myprovider.com.
 Execute “OK”.
5. Enter the digit collection timer values:
 Start Timer = 40 s.
 Short Timer = 5 s.
 Long Timer = 10 s.
 The digit map parameters are fully configured.
Select the CPS file Proceed as follows:

1. Navigate to the unit configuration dialogue:
 AP:/unit-19, Configuration - CPS.
2. Select a CPS:
 Select the previously downloaded CPS file from the drop down list.
 The correct CPS file is assigned now.
Create a PSTN port group Proceed as follows:

 AP:/unit-19, Configuration - Port Groups.
 Execute “Create PSTN Port Group…”.
2. Select the group index:
 Group Index = 5 (corresponding to the slot position of the PSTN line-
card; the matching is not required).
3. Select the number of ports:
 Number Of Ports = 16 (corresponding to the number of PSTN ports
on the corresponding PSTN linecard).
4. Execute “OK”.
 The PSTN port group is configured.
Create an ISDN-BA port Proceed as follows:

group 1. Navigate to the configuration dialogue:
 AP:/unit-19, Configuration - Port Groups.
 Execute “Create ISDN-BA Port Group…”.
2. Select the group index:
 Group Index = 12 (corresponding to the slot position of the ISDN-BA
linecard; the matching is not required).

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3. Select the number of ports:

 Number Of Ports = 16 (corresponding to the number of ISDN-BA
ports on the corresponding ISDN-BA linecard).
 The ISDN-BA port group is configured.
Configure the PSTN ports Proceed as follows:

 AP:/unit-19/portgroup-5/port-3, Configuration - Port PSTN.
2. Enable the subscriber port:
 Enable = true.
3. Enter a first subscriber identification. The first number will also be the
default number:
 Subscriber Number = 1001.
 Authorisation User Name = 1001.
 Authorisation Password = 1234.
 Display Name =
 Privacy = None.
 Execute “OK”
4. Configure the PSTN port parameters:
 Register As Global = true.
 Payphone = false.
 SIP Profile = none.
 Proxy / Registrar Profile = none.
 Codec / SDP Profile = none.
 PSTN Profile = none.
 Enterprise Profile = none.
6. Repeat the above steps for the PSTN port-8 with the subscriber number
1002.
 The PSTN ports are configured.
Configure the ISDN-BA ports Proceed as follows:

 AP:/unit-19/portgroup-12/port-2, Configuration - Port ISDN-BA.
2. Enable the subscriber port:
 Enable = true.
3. Enter a first subscriber identification. The first number will also be the
default number:
 Subscriber Number = 2001.
 Authorisation User Name = 2001.
 Authorisation Password = 1234.
 Display Name =
 Privacy = None.
 Execute “OK”

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4. Configure the ISDN-BA port parameters:

 Register As Global = true.
 Register Default Number >  Layer 1 Permanently Activated = false.
 SIP Profile = none.
 Proxy / Registrar Profile = none.
 Codec / SDP Profile = none.
 ISDN-BA Profile = none.
6. Repeat the above steps for the ISDN-BA port-5 with the subscriber num-
ber 2002.
 The ISDN-BA ports are configured.
Configure the PSTN linecard Proceed as follows:

1. On the PSTN linecard SUPM1, 16 user ports have to be configured and
brought into service. Please refer to the corresponding User Manual for
the detail configuration.
 The PSTN linecard is configured.
Configure the ISDN-BA line- Proceed as follows:

card 1. On the ISDN-BA linecard SUIQ1, 16 user ports have to be configured
and brought into service. Please refer to the corresponding User Manual
for the detail configuration.
 The ISDN-BA linecard is configured.
Create the cross connections Proceed as follows:

to the PSTN ports 1. Select the “Cross connections” view of the ECST.
 Click on the “Connection Wizard” button:
 The “Create TDM Connection” dialogue opens.

2. Set the connection parameters:
 Layer Rate = P0_nc.
 Directionality = Bidirectional.
 Protected = No.
 Number = 2.
 P0_nc, Multiple timeslots as one channel = 
 Label 1 = <anyName>.
3. Execute “Next ->”.
4. Select the Z-End CTPs:
 Select the IPSS4 unit, portgroup-5, port-3.
6. Select the A-End CTPs:
 Select the SUPM1 unit, port-3.
 Select the SUPM1 unit, port-8.

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7. Execute “Create”.
 The 2 connections from the SUPM1 ports to the ports of the port group 5
on the IPSS4 unit are created.
Create the cross connections Proceed as follows:

to the ISDN-BA ports
1. Select the “Cross connections” view of the ECST.
 Click on the “Connection Wizard” button:
 The “Create TDM Connection” dialogue opens.

2. Set the connection parameters:
 Layer Rate = P0-nc.
 Directionality = Bidirectional.
 Protected = No.
 Number = 3.
 P0_nc, Multiple timeslots as one channel = 
4. Select the Z-End CTPs:
 Select the IPSS4 unit, portgroup-12, icChannel.
6. Select the A-End CTPs:
 Select the SUIQ1 unit, icChannel.
 Select the SUIQ1 unit, port-2.
 Select the SUIQ1 unit, port-5.
7. Execute “Create”.
 The 2 connections from the SUIQ1 ports to the ports of the portgroup-12
on the IPSS4 unit and the connection from the SUIQ1 IC-channel to the
portgroup-12 IC-channel are created.
Configure the internal chassis Proceed as follows:

switch port on the COGE5 1. Navigate to the bridge VLAN configuration dialogue:
unit
 AP: /switching/bridges/bridge-1, VLANs.
 Click the button.
- The “Create a new VLAN” dialogue opens.
2. Enter the VLAN parameters:
 VLAN ID = 101.
 Name = <anyName>
4. Navigate to the bridge port configuration dialogue:
 AP: /switching/bridges/bridge-1, Ports.
 Select the chassis switch port the IPSS4 unit is connected to = AP: /
unit-11/iports/iport-19.

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5. Configure the port parameters:

 Mode = General.
 Acceptable Frametypes = Tagged
6. Select the “VLAN Configuration” tab:
 Execute “Add to VLANs”.
- The “Enter a Range of allowed VLANs” dialogue opens.
7. Enter the VLAN parameters:
 VLANs = 101.
 Egress Frametype = Tagged Frames
 The chassis switch port is configured.
Configure the Ethernet uplink Proceed as follows:

port on the COGE5 unit 1. Navigate to the bridge port configuration dialogue:
 AP: /switching/bridges/bridge-1, Ports.
 Select the port that is used as Ethernet uplink port = AP: /unit-11/
port-1.
2. Configure the port parameters:
 Mode = Trunk.
4. Navigate to the COGE5 SFP configuration dialogue:
 AP: /unit-11/port-1, Main - SFP.
 Configure the port type according to the plugged SFP or SFP+.
5. Navigate to the COGE5 physical configuration dialogue:
 AP: /unit-11/port-1, Main - Physical.
 Configure the Speed and Duplex parameter: Admin = Auto.
 Configure the flow control, Admin = .
6. Navigate to the COGE5 general configuration dialogue:
 AP: /unit-11/port-1, Main - General.
 Activate the interface: Admin Status = Up.
 The uplink port is configured.
Configure the NE synchroni- Proceed as follows:

zation 1. The MileGate can be synchronized via an external synchronization input
ESI-1 or ESI-2 on the COGE5 unit. Please refer to [314] User Guide
“TDM Services and Cross Connections” for the detail configuration.
 The NE synchronization is configured.
Save the configuration and Proceed as follows:

restart the IPSS4 unit 1. Save the configuration to the NE flash memory:
 Execute “Save to NE”.
2. Navigate to the equipment status dialogue:
 AP: /unit-19, Main - Equipment.
 Execute “Restart”.
 The SIP gateway is now configured and taken into service.
End of instruction

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6.4 Gateway configuration with equipment protection
6.4.1 Prerequisites
Before starting the commissioning of a SIP gateway with IPSS4 equipment

protection, the following prerequisites need to be fulfilled.
• The SIP gateway is fully configured as described in section 6.3 "Commis-
sioning of a gateway" (on page 98).
• A protecting IPSX4 unit with the same HW revision state as the working
IPSX4 unit is plugged in the MileGate subrack.
6.4.2 Configuration of the gateway
For the configuration of the SIP gateway with IPSS4 equipment protection,
the following steps have to be performed. The main steps are:
• Download the embedded software (ESW)
• Unassign the unit
• Create the protection group
• Save the EQP configuration
• Check the hardware compatibility
• Check the software compatibility
IPSX4 EQP protection This action list shows step by step how to configure the SIP gateway with
equipment protection.
Download and start the ESW Proceed as follows:

Download …):
2. In the table row of the protecting IPSS4 unit click on the “Software to
install” column.
3. Select the ESW from the drop down list.
4. In the table row of the protecting IPSS4 unit tick the “Delete unused Soft-
ware” box.
7. Click “OK”.
 The ESW is downloaded and the protecting IPSS4 unit starts with
the configured ESW.
 The protecting IPSS4 unit is now running with the correct ESW.
Unassign the protecting unit Proceed as follows:

1. Navigate to the unit handling dialogue of the protecting IPSS4 unit:
 AP: /unit-x, Main - Equipment.

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2. Unassign the unit:

 Execute “Unassign”.
 The protecting unit is unassigned.
Create the protection group Proceed as follows:

1. Navigate to the EQP dialogue on the working IPSS4 unit:
 AP: /unit-y, Configuration - EQP.
2. Create a protection group:
 Execute “Create EQP Group…”.
3. Configure the EQP group:
 Group Type = 1:1.
 Working Unit = <unit-y>.
 Protecting Unit = <unit-x>.
 The EQP group is configured.
Save the EQP configuration Proceed as follows:

1. Save the configuration to the NE flash memory:
 Execute “Save to NE”.
 The EQP configuration for the IPSX4 units is saved.
Check the hardware compati- Proceed as follows:

bility 1. Navigate to the EQP status dialogue on the working unit:
 AP: /unit-y, Status - EQP.
2. Check that the “HW compatible” column of the working unit is true.
3. Check that the “HW compatible” column of the protecting unit is true.
 The hardware of the working and protecting IPSX4 units are compatible.
Check the software compati- Proceed as follows:

bility
1. Navigate to the EQP status dialogue on the working unit:
 AP: /unit-y, Status - EQP.
2. Check that the “SW compatible” column of the working unit is true.
3. Check that the “SW compatible” column of the protecting unit is true.
 The softwares of the working and protecting IPSX4 units are compatible.
End of instruction
For the EQP detailed description please refer to section 5.7 "Equipment pro-
tection (EQP)" (on page 88).

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Operation User Manual
IPSS4
7 Operation
This section describes the operation functions of the IPSS4 unit.
In this section, you will find the following information:
• Port status parameters of the IPSS4 unit (section 7.1 "Port status man-
agement function" (on page 115)).
• Optical indicators description found on the IPSS4 unit front (section 7.2
"Unit optical indicators" (on page 117)).
• Indication and handling of possible faults on the IPSS4 unit (section 7.3
"Possible faults and related debugging" (on page 118)).
• Generic maintenance functions of the unit (section 7.4 "Maintenance" (on
page 124)).
Please note:
The operation functions described in this section assume a correctly config-
ured and operational IPSS4 unit.
7.1 Port status management function
Please note:
The maintenance functions for a subscriber port, e.g. locking and unlocking
a subscriber, are located on the corresponding service unit (linecard).
IPSS4 provides the status information only.
The following table gives some explanations about administrative and opera-
tional states according to ETS 300 376-1, Annex A:
Table 37: Administrative and operational states
Administrative Operational Explanation
state state
locked enabled The port has been locked by the management
system of the NE and there are no local fault
conditions.
disabled The port has been locked by the management
system of the NE and there is an NE fault.
unlocked enabled The port has been unlocked by the management
system of the NE and there is no NE fault.
The port is operational.
disabled The port has been unlocked by the management
system of the NE and there is an NE fault.
This state is also entered as part of the unblock-
ing procedure if the NE sent UNLOCK to the port
object and an acknowledgement from the signal-
ling needs to be awaited.
shutting down undefined The port state is awaiting the blocking from the
gateway after sending the block request.

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Only the ports in the state “unlocked” / “enabled” are operational in the
gateway. In all the other states, the associated user port is either blocked or
considered as blocked.

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7.2 Unit optical indicators
LEDs on the front of the IPSS4 unit are used to indicate to the operator the
alarm status summary of the unit and of the network traffic signals.
XXXXx R1B
37900374
UNIT TRAFFIC
Figure 31: Fault indication LEDs on the IPSS4 unit
Table 38: LED signalling on IPSS4

LED name Colour State Meaning
UNIT Red Failure The unit is not in service.
The unit is not able to provide the requested function due to
- equipment failure (total breakdown),
- mismatch of HW and SW.
Recovery from this error situation is done usually by replacement of the
unit HW or ESW.
Green / Red Booting or The unit has not been taken into service yet or the unit has not been
(blinking 1 Hz) waiting provisioned.
Recovery from this situation is done by taking the unit into service with
ECST.
Green Running The unit is up and running, it is ready to provide the required service.
Off Failure The system is not powered or outage of power supply on the unit or
outage of the LED.
TRAFFIC Red Failure One or more active failures on the unit, independent of the severity.
More detailed information is provided by ECST.
Off Normal Normal (error free) operation.

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IPSS4
7.3 Possible faults and related debugging
If a voice application has been created and configured but does not work,
please check the following hints to resolve the problem.
Please note:
The screen shots shown in the sections below are examples and might not
show the same contents as in your actual configuration.
7.3.1 MileGate active failures
• Check the ECST “Alarms” tab:

The active alarms window lists in a table all active failures of the whole
network element.
You find the failure description and the possible failure causes in the cor-
responding unit user manual or application user guide.
• If there are no active failures displayed in the resources used for the ser-
vice or application that is not performing as required, you need to check
for other possible causes. Refer to the sections below.
7.3.2 Unit not available (UNAV)
A reason for a service or application not performing as required can be a unit

that is not available, or a unit that is not assigned.
• Check the ECST “Tree View” tab:
All units required for the non-working service should be in the “Ok” state:
• If a unit is in the “Plugged” state, you have to assign the unit under the
AP: /unit-x, Main - Equipment or using the context menu of the unit:
• A valid embedded software (ESW) for the unit has to be downloaded and
configured under the AP: /unit-x, Main - Software or with the Software
Download tool of the ECST (Menu: Tools / Software Download …). For
details about the ESW installation refer to [355] User Manual “ECST”.
• Configure the unit and the corresponding cross connections.
7.3.3 PSTN or ISDN-BA user port down
• If you have not set the administrative state of a port to “Up”, the corre-
sponding MO will be fault free, but a voice service using this port will not
perform since the resources required for the service to work are inactive.

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IPSS4
Check that all the ports are in the administrative status “Up” using the MO
context menu as shown above or selecting the Admin And Oper Status
dialogue, e.g. AP: /unit-x/port-y, Main - Admin And Oper Status:
Set the administrative status to “Up”.

• Check that all the ports, logports and Pseudo Wires are in the opera-
tional status “Up” by selecting the Admin And Oper Status dialogue, e.g.
AP: /unit-x/port-y, Main - Admin And Oper Status:
If the operational state is “Down” make sure

− that the failure status of the MO has no active alarm (e.g. LOS or AIS).
• Check that the TDM cross connection is configured correctly according to
the application.
7.3.4 Wrong CPS on the IPSS4 unit
• Check that the configured CPS corresponds to the plugged gateway unit,
e.g. an IPSS4 unit requires a CPS prefixed with IPSS_, and that the CPS
also matches the network environment.
Please note:
There is not necessarily an active alarm in the MileGate if a CPS does not
match the unit or network environment.
• If there is a wrong custom parameter set in the IPSS4 unit configuration
selected, you have to
− download the required CPS to the MileGate with the CPS & Profile
tool under the ECST menu “Tools - CPS & Profile …”, if the CPS is
not available on the NE.
− apply the correct CPS to the IPSS4 unit under the AP: /unit-x, Config-
uration - CPS.

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IPSS4
Please note:
“Default” is generally not the CPS you can use in your network.
7.3.5 COGE5 Ethernet link down
If you have not set the administrative state of a COGE5 Ethernet port used
as uplink port to “Up”, the Ethernet port will be fault free, but a service using
this port will not perform since the resources required for the service to work
are inactive.
• Check that the Ethernet port on the COGE5 unit is in the AdminState
“Up” using the Ethernet port context menu
or selecting the AP: /unit-x/port-y, Main - Admin And Oper Status:
Set the administrative status to “Up”.

• Check that the Ethernet port on the COGE5 unit is in the operational sta-
tus “Up” by selecting the AP: /unit-x/port-y, Main - Admin And Oper Sta-
tus:

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IPSS4
If the operational state is “Down” make sure

− that the failure status of the Ethernet port has no active defect (e.g.
LOS or ENA).
• Check that the switch or router in the aggregation network is in the “Up”
state, and that correct Ethernet cables or fibres are connected, depend-
ing on the PHY type.
• Check that the electrical or optical COGE5 Ethernet port is configured
correctly as a VLAN trunk port in the switching view under the AP: /
switching/bridges/bridge-1/ports.
• Check if the link comes up when configuring a fixed PHY speed instead
of auto-negotiation.
7.3.6 Service VLAN configuration
If you have not configured the MileGate internal port between the IPSS4 unit
and the COGE5 unit correctly, the service using this port will not perform.
• Check that the internal COGE5 Ethernet port is configured correctly as a
VLAN trunk port or general port in the switching view under the AP: /
switching/bridges/bridge-1/ports.

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IPSS4
• Check that the internal COGE5 Ethernet port has the correct VLAN
assigned and the correct Egress Frametype in the switching view under
the AP: /switching/bridges/bridge-1/ports/status. The VLAN must be the
same as the configured VLAN of the network interface on the IPSS4 unit.
The Egress Frametype must be “Tagged Frames”.
Internal COGE5 port,

assigned VLAN IDs
IPSS4 network interface,

configured VLAN ID
• Check if the configured service VLAN ID for the respective service is the
one used in the aggregation network.
• Check if the single tag configuration corresponds with the aggregation
network.
7.3.7 Softswitch not reachable
• If the connection to the registrar or proxy server is not working, check the
network connection with the IPSS4 built in “ping” feature under the AP: /
unit-x, Status - Gateway.
• Click on the “Ping…” button.
• Enter the IP address of the registrar or proxy server:
• Click the “OK” button.
• Check the number of echo responses to be the same as the number of
echo requests.
Please note:
If the proxy is not available, the IPSS4 unit is still able to route local calls if
the LCR feature is enabled.

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IPSS4
7.3.8 TDM connections not correct
• Check that the TDM connections are configured correctly according to

the application.
• The connections and the static and dynamic CTPs are visible in tabular
form in the ECST “Cross Connections” view.
Make sure that all configuration parameters are set as required.
• The connections parameters are visible in tabular form in the right part of
the ECST “Cross Connections” view.
Make sure that all configuration parameters are set as required.
Please note:
The SUPM1, SUPM5, SUIx1, and IPSS4 units use fixed CTPs which have
not to be configured.

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IPSS4
7.4 Maintenance
7.4.1 Inventory data
It is possible to read inventory data from the IPSS4 unit via the ECST at the
following access point:
AP: /unit-x, Main - Inventory.
7.4.2 Unit ESW download
It is possible to update the embedded software (ESW) of the IPSS4 unit via
software download.
Please refer to [355] User Manual “ECST” for the description of the ESW
download.

The assignment of new embedded software restarts the IPSS4 unit.
→ Thus, the installation of new ESW on the unit affects all traffic func-
tions of the IPSS4 unit.
7.4.3 ESW upgrade with EQP
When upgrading the ESW on 1:1 equipment protected IPSS4 units, care
must be taken concerning the traffic interruptions and which unit will finally
be the active unit. At the end of the upgrade procedure the working unit shall
be the active unit.
It is assumed that the working unit is plugged in slot 19 and the protecting
unit is plugged in slot 21 of the MileGate 2510 subrack.
ESW upgrade procedure 1 The following procedure provides the upgrade process with one traffic inter-
ruption of about 85 s.
Isolate the working unit Proceed as follows:

1. Isolate the working IPSS4 unit:
 AP: /unit-19, Status – EQP.
 Execute the “Isolate Unit” command.
 The working IPSS4 unit is isolated, i.e. it will not perform a protection
switchover.

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IPSS4
ESW download and start Proceed as follows:

Download …):
2. In the table row of the working IPSS4 unit click on the “Software to
install” column.
3. Select the new ESW from the drop down list.
install” column.
6. In the table row of the working and protecting IPSS4 units tick the
“Delete unused Software” box.
9. Click “OK”.
 The ESW is downloaded and the IPSS4 units start with the config-
ured ESW.
 Traffic will be interrupted for about 85 s.
 The new ESW is active on the working and the protecting IPSS4 units.
The working IPSS4 unit in slot 19 remains the active unit.
Join the working unit Proceed as follows:

1. Join the working IPSS4 unit:
 AP: /unit-19, Status – EQP.
 Execute the “Join Unit” command.
 The working IPSS4 unit is able again to perform a protection switchover.
End of instruction
ESW upgrade procedure 2 An alternative procedure requires two shorter interruptions of about 20 s
instead of one long interruption.
ESW download Proceed as follows:

Download …):
install” column.
install” column.
6. In the table row of the working and protecting IPSS4 units tick the
“Delete unused Software” box.
7. Execute the “Download” command.
 The new ESW is downloaded to the working and protecting IPSS4 units.

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IPSS4
ESW upgrade on the protect- Proceed as follows:

ing unit
Download …):
install” column.
4. Press the “Start …” button.
5. In the “Parameters for command Start” select the algorithm “Upgrade
Units Only”.
6. Click “OK”.
 The protecting IPSS4 unit starts with the configured ESW.
 This takes about 85 s.
 The IPSS4 unit is now running with the correct ESW.
Manual switch to protecting 1. Perform a manual switch-over from the working to the protecting IPSS4
unit unit:
 AP: /unit-19, Status - EQP
 Execute the “Manual Switch-Over” command.
 Traffic will be switched to the protecting unit.
2. Wait until the working unit has rebooted.
ESW upgrade on the working 1. Open the Software Download tool of the ECST (Menu: Tools / Software
unit Download …):
install” column.
4. Press the “Start …” button.
5. In the “Parameters for command Start” select the algorithm “Upgrade
Units Only”.
6. Click “OK”.
 The working IPSS4 unit starts with the configured ESW.
Manual switch to working unit 1. Perform a manual switch-over from the protecting to the working IPSS4
unit:
 AP: /unit-19, Status - EQP
 Execute the “Manual Switch-Over” command.
 Traffic will be switched to the working unit.
2. Wait until the protecting unit has rebooted.
 The ESW upgrade is complete.
End of instruction

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User interface reference User Manual
IPSS4
8 User interface reference

This section gives a complete reference of the managed objects, properties,
and commands of the IPSS4 functional unit of the MileGate.
In this section, you will find the following information:
• An introduction (section 8.1 "Introduction" (on page 127)),
• Profile configuration parameters (section 8.2 "Profiles" (on page 131)),
• Unit management commands and parameters (section 8.3 "AP: / unit-x:
IPSS4" (on page 139)),
• Control and media parameters (sections 8.4 "AP: / unit-x / control" (on
page 171) and 8.5 "AP: / unit-x / media" (on page 172)),
• Hunting groups commands and parameters (sections 8.6 "AP: / unit-x /
huntgroups" (on page 173) and 8.7 "AP: / unit-x / huntgroups / group-a
(ISDN-BA)" (on page 175)),
• Portgroup commands and properties (section 8.8 "AP: / unit-x / port-
group-y (PSTN) or (ISDN-BA)" (on page 184)).
• Port and IC-channel management commands and parameters (sections
8.9 "AP: / unit-x / portgroup-y / port-z (PSTN)" (on page 186), 8.10 "AP: /
unit-x / portgroup-y / port-z (ISDN-BA)" (on page 194) and 8.11 "AP: /
unit-x / portgroup-y / icChannel" (on page 202)).
Please note:
ISDN-BA subscriber units and the corresponding ISDN-BA applications will
be available in a future release.
For a description on how to configure and bring into operation the IPSS4 unit
and its main functions, please refer to section 6 "Commissioning" (on
page 93).
8.1 Introduction
Below, you will find a detailed description of all the configuration parameters
and operations belonging to the managed objects model (MOM) for the
IPSS4 service unit.
The Figure 32 shows the access point (AP) tree for the IPSS4 unit with its
managed objects.

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IPSS4
<ap >
MileGate
0...2 <ap >

unit-x: IPSX4
1 <ap>
control
1 <ap>
media
1 <ap>
huntgroups
0 ... 304 <ap>

group-a
<ap>
0 ... 50
portgroup-y:
PSTN
0 ... 64 <ap>
port-z
<ap>
0 ... 20
portgroup-y:
ISDN-BA
1 <ap>
icChannel
0 ... 16 <ap>
port-z
Figure 32: MOM (managed object model) of the IPSS4 unit
With these managed objects (MOs) the following functions are covered:

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IPSS4
Table 39: Managed objects (MOs) for IPSS4

MO Description of the management functions
unit-x: IPSX4 Restart of the unit, management of the unit ESW, labelling,
(ipss4_r2a) indication of equipment status, read of inventory data, access
to logbooks.
Gateway:
- Management of the media path with jitter buffer configuration,
RTP port range.
- SIP configuration and supplementary services management.
- Configuration of the IP address and the DSCP for media and
control flows.
Proxy, registrar and DNS servers:
- Configuration of the addresses and the registration parame-
ters.
Digit map:
- Configuration of the digit map and the digit collection timers.
CPS:
- Selection of the CPS (custom parameter set) files.
EQP:
- Creation and deletion of an equipment protection group. EQP
status display.
Port groups:
- Adding and deleting of port groups.
Fault management and performance management.
Unit status.
control This AP has no function in the context of this MileGate release.
media This AP has no function in the context of this MileGate release.
huntgroups Creation and deletion of trunk hunting groups for ISDN-BA
ports.
group-a Adding and deleting ports to or from a trunk hunting group.
Configuration of the ISDN-BA user ports of the corresponding
user port units.
portgroup-y: PSTN Creation and deletion of PSTN ports.
port-z: PSTN Configuration of the PSTN user port.
Port status.
portgroup-y: ISDN- Creation and deletion of ISDN-BA ports.
BA
icChannel Configuration and fault management of the IC channel.
port-z: ISDN-BA Configuration of the ISDN-BA user port.
Port status.
For each of the managed objects, properties and commands, the ECST
“Tree Views” are given.
This reference section comprises the management functions:
• Overview,
• Main,
• Configuration,
• Fault Management,
• Performance Management, and
• Status.
Most of the APs only offer a part of the management functions listed above.

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IPSS4
The order of appearance of the management function descriptions is in

accordance with the APs in the ECST AP tree and the availability of the
management functions of each AP.
In the tables of the sections below, the parameter default values for proper-
ties are underlined.
Please note:
For better legibility of numbers in this user guide, inverted commas are used
when the number’s size exceeds three digits (e.g. 40’000). In parameter
entry fields of the ECST, these inverted commas must not be entered.
Instead, the numbers are entered without these inverted commas (e.g.
40000).
Please note:
Screenshots presented in this reference are examples and show configura-
tions or data that may not correspond to the view you see when managing
your MileGate equipment.

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IPSS4
8.2 Profiles
Profiles are created with the ECST GUI. For a detailed description of the pro-
file creation and download procedures to the NE please refer to [355] User
Manual “ECST”.
Please note:
The profile templates are only available if the ESW of the IPSS4 unit has
been imported into the ECST.
8.2.1 SIP profile
Table 40: SIP profile

Operation Name Parameter Name Range Description / Details
Profile Templates Template SipProfile_1.00.00 Select the SIP profile template as a base for the
parameter editing.
Profile, Information Name 3 … 64 characters User editable name for the profile.
Valid characters are a-z, A-Z, 0-9 and _.
Description 0 … 64 characters User editable description for the profile.
Profile, Data, SIP Home Domain 0 … 63 characters Domain name present in the URI address-of-
Profile record.
The default home domain is empty.
SIP Port Number 1 … 5060 … 9999 UDP port number of the SIP protocol.
Each profile used on an IPSS4 unit is

required to have a different SIP port number.
Country Code 0 … 8 characters Enter the country code including the leading “+”
sign.
The default country code is +41.
Area Code 0 … 8 characters Enter the area code excluding the leading “0”
character.

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8.2.2 SIP Proxy / Registrar profile
Table 41: SIP Proxy / Registrar profile

Profile Templates Template SipProxyRegistrarProfile Select the SIP Proxy / Registrar profile template
_1.04.00 as a base for the parameter editing.
Profile, Data, SIP Proxy Mode Primary Only Only a primary proxy is configured
Proxy / Registrar
Revertive IPSS4 switches to the secondary proxy if the pri-
Profile, Proxy
mary proxy fails and switches back to the pri-
mary proxy as soon as the primary proxy
becomes available again.
Non Revertive IPSS4 switches to the secondary proxy if the pri-
mary proxy only if the secondary proxy fails and
the primary proxy is available again.
DNS / RFC 3263 No proxy is configured. The proxy is found via
DNS queries.

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Table 41: SIP Proxy / Registrar profile (continued)

Profile, Data, SIP Proxy Address <IPv4 address> Outbound proxy address as IP or fully qualified
Proxy / Registrar e.g. 172.31.63.181 domain name (FQDN).
Profile, Proxy, Pri- A proxy address is required when no DNS server
<FQDN>
mary Proxy address is configured. If no proxy is used the
e.g. proxy.london.com proxy address configuration must be empty and
the proxy port configuration must be 0.
Please refer to section 5.5 "Proxy server and
registrar server functions" (on page 77) for fur-
ther information.
The default address is empty.
Proxy Port 0 … 5060 … 9999 Outbound proxy port number.
Profile, Data, SIP Proxy Address <IPv4 address> Outbound proxy address as IP or fully qualified
Proxy / Registrar e.g. 172.31.63.181 domain name (FQDN).
Profile, Proxy, Sec- A proxy address is required when no DNS server
<FQDN>
ondary Proxy address is configured. If no proxy is used the
e.g. proxy.london.com proxy address configuration must be empty and
ther information.
Proxy Port 0 … 9999 Outbound proxy port number.
Profile, Data, SIP Enabled  Enable the proxy availability for this gateway.
Proxy / Registrar 
Profile, Proxy,
Proxy Availability Method OPTIONS Send an OPTIONS or a REGISTER request to
Check the proxy to check the availability of the proxy.
REGISTER
Interval 10 … 100 s Enter the interval on which the proxy availability
check shall be performed.
Note that during proxy unavailability the check
interval is 10 s, irrespective of the configuration
value.
Profile, Data, SIP Registrar Address <IPv4 address> Registrar address configured as IP or fully quali-
Proxy / Registrar e.g. 172.31.63.181 fied domain name (FQDN).
Profile, Registrar A registrar address is always required even if the
<FQDN>
registration mode is set to no registration. Please
e.g. regist.london.com refer to section 5.5 "Proxy server and registrar
server functions" (on page 77) for further infor-
mation.
Registrar Port 0 … 5060 … 9999 Registrar port number.
Registration Mode No Registration Registration mode with a registrar.
“No Registration” is used if all subscribers are
One-By-One Registration
registered by provisioning in the registrar.
Registration Expira- 1 … 100 h Enter the registration expiration time.
tion Time

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8.2.3 SIP Codec / SDP profile
Table 42: SIP Codec / SDP profile

Profile Templates Template SipCodecSdpProfile Select the SIP Codec / SDP profile template as a
_1.01.00 base for the parameter editing.
Profile, Data, SIP G.711 Silence Sup-  Enable the silence suppression function with the
Codec / SDP Pro- pression  G.711 codec.
file, Codec / SDP,
Voice G.729AB Silence  Enable the G.729AB silence suppression.
Suppression 
Profile, Data, SIP T.38 Support  If enabled all G3 fax formats apart from V.34 are
Codec / SDP Pro-  converted to T.38, if the partner gateway sup-
file, Codec / SDP, ports the T.38 relay function. V.34 fax is trans-
SDP Advanced ported as VBD (Voice Band Data).
DTMF Relay Sup-  If enabled DTMF tones are converted to
port  RFC2833 packets or transported as SIP INFO
messages. Inband transport of DTMF tones is
squelched.
Telephone-Event 96 … 127 The RTP payload format for the DTMF transport
Payload Type (=telephone-event) uses a dynamic payload
type.
Clearmode Pay- 96, 97 … 127 The RTP payload format for the clearmode data
load Type transport uses a dynamic payload type.

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Table 42: SIP Codec / SDP profile (continued)

DTMF Relay Type RFC 2833 Defines, if the DTMF tones are sent according to
RFC 2833 or by SIP INFO messages.
SIP INFO
Only relevant if DTMF Relay Support is enabled.
Profile, Data, SIP Voice Codec 1 None Voice codec 1 is the first codec offered in the
Codec / SDP Pro- outgoing SDP message.
G.711 A
file, Codec / SDP, Select the voice codec to be used.
Voice Codec Selec- G.729 AB
tion Voice codec 2 is the second codec offered in the
Voice Codec 2 None
outgoing SDP message.
G.711 A
Select the voice codec to be used.
G.729 AB
Voice Codec 3 None Voice codec 3 is the third codec offered in the
G.711 A
G.729 AB
8.2.4 SIP PSTN port profile
Table 43: SIP PSTN port profile

Profile Templates Template SipPstnProfile_2.02.02 Select the SIP PSTN port profile template as a
base for the parameter editing

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IPSS4
Table 43: SIP PSTN port profile (continued)

Profile, Data, SIP Hotline  Enable or disable the hotline supplementary ser-
PSTN Profile, Sup-  vice.
plementary Ser-
vices Per Call CLIR  Enable or disable the per call calling line ID
 restriction (CLIR) supplementary service.
Calling Line ID  Enable or disable the calling line ID presentation

Presentation  (CLIP) supplementary service.
CLIP Mode FSK Transmission Physical layer data transmission mode for CLIP:
FSK or DTMF
DTMF Transmission
Service Control On Gateway The supplementary services listed under “Ser-
vices (Gateway Control)” below can be con-
On Proxy
trolled by the gateway or by the proxy. If proxy is
selected the services configuration below is not
relevant.
Profile, Data, SIP Call Waiting  Enable or disable the call waiting supplementary
PSTN Profile, Sup-  service.
plementary Ser-
vices, Services Enquiry Call  Enable or disable the enquiry call supplementary
(Gateway Control)  service.
3-Way Call  Enable or disable the 3-way call supplementary

 service.
Attended Call  Enable or disable the attended call transfer sup-

Transfer  plementary service.
Profile, Data, SIP Call Forwarding  Enable or disable the call forwarding uncondi-
PSTN Profile, Sup- Unconditional  tional supplementary service.
plementary Ser-
vices, Network Call Forwarding  Enable or disable the call forwarding busy sup-
Services Busy  plementary service.
Call Forwarding No  Enable or disable the call forwarding uncondi-

Reply  tional supplementary service.
Please note:
Refer to section 5.4.3 "Supplementary services implemented for PSTN sub-
scribers" (on page 60) for further information about the supplementary ser-
vices.

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IPSS4
8.2.5 SIP Enterprise profile
Table 44: SIP Enterprise profile

Profile Templates Template SipEnterpriseProfile_1.00. Select the SIP Enterprise profile template as a
00 base for the parameter editing
Profile, Data, SIP Off-Net Prefix 0 … 3 characters Enterprise prefix for externals calls.
Enterprise Profile, External calls are signalled to a PSTN subscriber
PSTN Enterprise, with a specific dial tone, controlled by the CPS.
External Dial Tone
Profile, Data, SIP Prefix 0 … 5 characters Enterprise blocked prefixes for PSTN subscrib-
Enterprise Profile, ers.
PSTN Enterprise, Enter blocked prefixes e.g. 00 to block interna-
Blocked Number tional calls.
Prefixes
Add… Add a prefix to be blocked.
Up to 3 prefix entries are configurable.
Only available with the ECST.
Remove Delete a selected prefix.
Only available with the ECST
Profile, Data, SIP Enabled  Enable or disable the enterprise hotline number
Enterprise Profile,  service.
PSTN Enterprise,
Hotline Hotline Number 0 … 20 characters Enter the enterprise hotline number.
The default hotline number is empty
Start Timer 0…5…9s Time after the off-hook event to automatically
dial the hotline number.

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IPSS4
Please note:
Refer to section 5.4.3 "Supplementary services implemented for PSTN sub-
scribers" (on page 60) for further information about the PSTN supplementary
services.
Please note:
Refer to section 5.4.4 "Enterprise supplementary services implemented for
PSTN subscribers" (on page 68) for further information about the enterprise
supplementary services.
8.2.6 SIP ISDN-BA port profile
Table 45: SIP ISDN-BA port profile

Profile Templates Template SipIsdnBaProfile_2.01.00 Select the SIP ISDN-BA port profile template as
a base for the parameter editing
Profile, Data, SIP Hotline  Enable or disable the hotline supplementary ser-
ISDN-BA Profile,  vice.
Supplementary Ser-
vices Per Call CLIR  Enable or disable the per call calling line ID
Profile, Data, SIP Call Forwarding  Enable or disable the call forwarding uncondi-
ISDN-BA Profile, Unconditional  tional supplementary service.
Supplementary Ser-
vices, Network Ser- Call Forwarding  Enable or disable the call forwarding busy sup-
vices Busy  plementary service.

Refer to section 5.4.5 "Supplementary services implemented for ISDN-BA

subscribers" (on page 69) for further information about the supplementary
services.

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IPSS4
8.3 AP: / unit-x: IPSS4
8.3.1 AP: / unit-x, Overview
For a description of the

− “Overview - Alarms”, and
− “Overview - Cross Connections”
management functions, please refer to [302] User Guide “MileGate 2310/
2510”.
8.3.2 AP: / unit-x, Main

− “Main - General”,
− “Main - Equipment”,
− “Main - Inventory”, and
− “Main - Software”
2510”.
8.3.2.1 AP: / unit-x, Main - Logbooks
The logbook entries stored on the unit are presented in different logbooks.
For the general view on a logbooks dialogue of the ECST, refer to [355] User
Manual “ECST”.
For a detailed explanation of the ECST logbooks commands and properties
also refer to [355] User Manual “ECST”.
For a description of the alarm, event and equipment logbooks please refer to
[355] User Manual “ECST”.
The following logbook is specific on the IPSS4 unit:
Table 46: AP: / unit-x, Main - Logbooks

Logbook Logbook parameter Description
Protocol Error Time Stamp Date and time of the event
Format: YYYY-MM-DDThh:mm:ss
Description This parameter provides a description of unexpected events as
- PSTN call control failures,
- ISDN call control failures,
- SIP call control failures, or
- DNS failures, e.g. “no such name” query responses.
Note that DNS access failures will raise an alarm and will not
be written to the protocol error logbook.

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IPSS4
8.3.3 AP: / unit-x, Configuration
8.3.3.1 AP: / unit-x, Configuration - Gateway - SIP
Table 47: AP: / unit-x, Configuration - Gateway - SIP

SIP Gateway Name 0 … 63 characters User editable name allocated to the gateway.
The default gateway name is “gateway”.
Home Domain 0 … 63 characters Domain name present in the URI address-of-
record.
The default home domain is empty.
SIP Port Number 1 … 5060 … 9999 UDP port number of the SIP protocol.
Country Code 0 … 8 characters Enter the country code including the leading “+”
sign.
The default country code is +41.
If no country code is required this field might be
left empty.
Area Code 0 … 8 characters Enter the area code excluding the leading “0”
character.
If no area code is required this field might be left
empty.
Retransmission 100 … 500 … 2’000 ms Retransmission timer T1, i.e. the round-trip time
Timer T1 RTT estimate.
Max Retransmis- 1 … 4 … 10 s Maximum retransmission interval for non-INVITE
sion Interval T2 requests and INVITE responses.
SIP Extension  Support of the reliable provisional response
100rel Required  message delivery (RFC3262).

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IPSS4
Table 47: AP: / unit-x, Configuration - Gateway - SIP (continued)

Asserted Identity None Private Extensions to SIP for Asserted Identity
Mode within trusted networks.
Asserted
Refer also to section 5.4.2.8 "Asserted identity
Preferred and privacy" (on page 53).
Overlap Signalling  Overlap signalling uses multiple INVITEs for
 consecutive dialled digits instead of en-bloc sig-
nalling.
Refer also to section 5.4.2.9 "Outgoing overlap
signalling" (on page 55).
Overlap T10 Timer 1 … 10 … 60 s Call setup expiry timer T10 timer used with over-
lap signalling.
Refer also to section 5.4.2.9 "Outgoing overlap
signalling" (on page 55).
SIP, Session Timer UAC Request Ses-  The gateway requests the use of session timer
sion-Timer  for the outbound calls.
UAS Request Ses-  The gateway requests the use of session timer
sion-Timer  for the inbound calls.
Session Expiration 180 … 1’800 … 86’400 s Specifies the time at which the session is consid-
ered to be timed out.
Please note:
Changes in the parameters below cause a re-creation of all affected ports,
i.e. the ports are removed and the respective subscriber numbers are de-
registered. The ports are thereafter added again and the respective sub-
scriber numbers are newly registered.
→ Home Domain,
→ SIP Port Number,
→ Country Code,
→ Area Code.

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IPSS4
8.3.3.2 AP: / unit-x, Configuration - Gateway - Supplementary Services - Available Services
Add a new entry to the Blocked Number Prefixes table:

207
IPSS4
Table 48: AP: / unit-x, Configuration - Gateway - Supplementary Services - Available Services

PSTN/ISDN-BA Hotline  Enable or disable the hotline supplementary ser-
 vice.
Per Call CLIR  Enable or disable the per call calling line ID
PSTN/ISDN-BA, Call Forwarding  Enable or disable the call forwarding uncondi-

Network Services Unconditional  tional supplementary service.
Call Forwarding  Enable or disable the call forwarding busy sup-

Busy  plementary service.

PSTN Calling Line ID  Enable or disable the calling line ID presentation

Presentation  (CLIP) supplementary service.
CLIP Mode, FSK Transmission Physical layer data transmission mode for CLIP:
FSK or DTMF
DTMF Transmission
Service Control On Gateway The supplementary services below can be con-
On Proxy trolled by the gateway or by the proxy. If proxy is
selected the services configuration below is not
relevant.
PSTN, Services Call Waiting  Enable or disable the call waiting supplementary
(Gateway Control)  service.
Enquiry Call  Enable or disable the enquiry call supplementary

 service.
3-Way Call  Enable or disable the 3-way call supplementary

 service.
Attended Call  Enable or disable the attended call transfer sup-

Transfer  plementary service.
PSTN Enterprise, Off-Net Prefix 0 … 3 characters Enterprise prefix for externals calls.
External Dial Tone External calls are signalled to a PSTN subscriber
with a specific dial tone, controlled by the CPS.
PSTN Enterprise, Prefix 0 … 5 characters Enterprise blocked prefixes for PSTN subscrib-
Blocked Number ers.
Prefixes Enter blocked prefixes e.g. 00 to block interna-
tional calls.
Add… Add a prefix to be blocked.
Up to 3 prefix entries are configurable.
Remove Delete a selected prefix.
PSTN Enterprise, Enabled  Enable or disable the enterprise hotline number
Hotline  service.
This service is only available for PSTN subscrib-
ers.
Hotline Number 0 … 20 characters Enter the enterprise hotline number.
The default hotline number is empty
Start Timer 0…5…9s Time after the off-hook event to automatically
dial the hotline number.

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Table 48: AP: / unit-x, Configuration - Gateway - Supplementary Services - Available Services (con-
tinued)
Subscriber Time Enabled  Enable or disable the daylight saving time cor-
Correction, Euro-  rection according to the European standard:
pean Standard The time offset is 1 hour.
Daylight saving starts the last Sunday of March,
at 01h00 AM.
Daylight saving ends the last Sunday of October,
at 01h00 AM.
Subscriber Time Enabled  Enable or disable the daylight saving time cor-
Correction, Non-  rection according to the configuration parame-
European Standard ters below.
Note that only one of the two daylight saving
time correction modes can be enabled at a time.
Time Offset 0 … 12 h Time offset for the daylight saving
Start Date And DD MM YYYY hh mm ss Date and time when the daylight saving time
Time starts.
Stop Date And Time DD MM YYYY hh mm ss Date and time when the daylight saving time
ends.
Please note:
A change in the parameter below causes a re-creation of all affected ports,
i.e. the respective subscriber numbers are de-registered and newly regis-
tered.
→ Service Control.

207
IPSS4
8.3.3.3 AP: / unit-x, Configuration - Gateway - Supplementary Services - Service Codes

207
IPSS4
Table 49: AP: / unit-x, Configuration - Gateway - Supplementary Services - Service Codes

Invocation Codes Toggle Or Accept 0…2…8 Invocation code for accepting a call or toggling
(PSTN)a Call between two calls.
Reject Call 0…8 Invocation code for rejecting a call.
Disconnect Call 0…1…8 Invocation code for disconnecting a call.
Transfer Call 0…4…8 Invocation code for transferring a call.
3-Way Call 0…3…8 Invocation code for the setup of a 3-way call.
Call Waiting Activate 0 … 16 characters Subscriber code to activate the service.
(PSTN)b The default value is *43#.
Deactivate 0 … 16 characters Subscriber code to deactivate the service.
The default value is #43#.
Initially Activated  The supplementary service Call Waiting
 becomes automatically activated when the port
is added or the port configuration is changed.
The port profile must have the call waiting sup-
plementary service enabled.
Hotline (PSTN/ Activation, Activate 0 … 16 characters Subscriber code to enter the hotline number.
ISDN-BA)c The default value is *53*.
Activation, Termina- 0 … 1 characters Subscriber code to terminate the hotline number
tion Character entry.
The default value is #.
Deactivate 0 … 16 characters Subscriber code to deactivate the service.
The default value is #53#.
Interrogate 0 … 16 characters Subscriber code to check the active hotline num-
ber.
The default value is *#53#.
Start Timer 1…5…9s Time value after going off hook to dial the hotline
number.
Per Call CLIR Activate 0 … 16 characters Subscriber code to activate the service and sub-
(PSTN/ISDN-BA) c sequently entering the subscriber number.
The default value is *31*.
Termination Char- 0 … 1 characters Subscriber code to terminate the indication
acter restricted number entry.
The default value is #.
Call Diversion Activation, Activate 0 … 16 characters Subscriber code to enter the call forwarding
(PSTN/ISDN-BA), number, e.g. *21*.
Call Forwarding The default value is empty.
Unconditional Subscriber code to terminate the call forwarding
Activation, Termina- 0 … 1 characters
tion Character number entry, e.g. #.
The default value is empty.
Deactivate 0 … 16 characters Subscriber code to deactivate the service, e.g.
#21#.
Interrogate 0 … 16 characters Subscriber code to check the active call forward-
ing number.

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IPSS4
Table 49: AP: / unit-x, Configuration - Gateway - Supplementary Services - Service Codes (contin-
ued)
Call Forwarding The default value is empty.
Busy Subscriber code to terminate the call forwarding
#63#.
ing number.
Call Forwarding No The default value is empty.
Reply Subscriber code to terminate the call forwarding
#61#.
ing number.
a. Invocation codes are applied by the PSTN subscriber by pressing the hook-flash button (R-button), waiting for the recall
dial tone (rdt) and then pressing the desired invocation code button. Please refer to section 5.4.3 "Supplementary ser-
vices implemented for PSTN subscribers" (on page 60) for further information.
b. Supplementary service for PSTN subscribers. Please refer to section 5.4.3 "Supplementary services implemented for
PSTN subscribers" (on page 60) for further information.
c. Supplementary service for PSTN and ISDN-BA subscribers. Please refer to section 5.4.3 "Supplementary services im-
plemented for PSTN subscribers" (on page 60) for further information.

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IPSS4
8.3.3.4 AP: / unit-x, Configuration - Gateway - Media - Codec / SDP
Table 50: AP: / unit-x, Configuration - Gateway - Media - Codec / SDP

Codec / SDP, Voice G.711 Silence Sup-  Enable the silence suppression function with the
pression  G.711 codec.
G.729AB Silence  Enable the G.729AB silence suppression.

Suppression 
Codec / SDP, SDP T.38 Support  If enabled all G3 fax formats apart from V.34 are
Advanced  converted to T.38, if the partner gateway sup-
ports the T.38 relay function. V.34 fax is trans-
ported as VBD (Voice Band Data).
DTMF Relay Sup-  If enabled DTMF tones are converted to
port  RFC2833 packets or transported as SIP INFO
messages. Inband transport of DTMF tones is
squelched.
Telephone-Event 96 … 127 The RTP payload format for the DTMF transport
Payload Type (=telephone-event) uses a dynamic payload
type.
Clearmode Pay- 96, 97 … 127 The RTP payload format for the clearmode data
load Type transport uses a dynamic payload type.
DTMF Relay Type RFC 2833 Defines, if the DTMF tones are sent according to
RFC 2833 or by SIP INFO messages.
SIP INFO
Only relevant if DTMF Relay Support is enabled.
Codec / SDP, Voice Voice Codec 1 None Voice codec 1 is the first codec offered in the
Codec Selection outgoing SDP message.
G.711 A
G.729 AB
Voice Codec 2 None Voice codec 2 is the second codec offered in the
G.711 A
G.729 AB

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IPSS4
Table 50: AP: / unit-x, Configuration - Gateway - Media - Codec / SDP (continued)
Voice Codec 3 None Voice codec 3 is the third codec offered in the
G.711 A
G.729 AB
8.3.3.5 AP: / unit-x, Configuration - Gateway - Media - Jitter Buffer / RTP
Table 51: AP: / unit-x, Configuration - Gateway - Media - Jitter Buffer / RTP
Jitter Buffer / RTP, Nominal Threshold 0 … 50 … 200 ms Nominal fill level for voice packets.
Adaptive Jitter step 5 ms
Buffer
Fax+Clearmode 0 … 80 … 200 ms Nominal fill level for Fax and clearmode packets.
Threshold step 5 ms
Low Threshold 0 … 200 ms The low threshold must be = the nominal and
step 5 ms Fax threshold.
High Threshold 50 … 200 ms The high threshold must be = the nominal and
step 5 ms Fax threshold.
Response Time 0.0 … 2.5 s Dynamic behaviour of the fill level adaptation.
step 0.1 s
Jitter Buffer / RTP, Start Port Number 10’000 … 50’000 … 65’535 Begin of the RTP port number range that is used
RTP Port Range for media connections.
Stop Port Number 13’000 … 54’000 … 65’535 End of the RTP port number range that is used
for media connections.

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IPSS4
8.3.3.6 AP: / unit-x, Configuration - Gateway - Network Interface
Table 52: AP: / unit-x, Configuration - Gateway - Network Interface

IP Gateway IP <IPv4 address> The same IP address is used for both the control
Address e.g. 172.31.63.99 exchange with the softswitch and the RTP
stream.
Any change on IP parameters requires the con-
figuration to be saved, and to restart the unit. a
The default gateway IP address is empty.
Subnet Mask <IPv4 address> Subnet mask of the gateway.
e.g. 255.255.255.0 The default subnet mask is empty.
Default Gateway <IPv4 address> The same default gateway is used for both the
e.g. 172.31.63.10 control exchange with the softswitch and the
RTP stream.
The default IP address of the default gateway is
empty.
DNS Server Preferred DNS <IPv4 address> IP address of the preferred DNS server.
Address Server e.g. 172.31.63.181 A domain name system (DNS) server address is
required when the proxy address is configured
as fully qualified domain name (FQDN) or when
no proxy address is configured.
If no DNS server is used the configuration must
be empty.
ther information.
The default preferred DNS server IP address is
empty.
Alternate DNS <IPv4 address> IP address of the alternate DNS server.
Server e.g. 172.31.63.182 The default alternate DNS server IP address is
empty.

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Table 52: AP: / unit-x, Configuration - Gateway - Network Interface (continued)

QoS/DiffServ DSCP Control SIP 0 … 40 … 63 Class of service for SIP packets.
Class of service for IP packets marked in the IP
header. This can be used to prioritise these
packets over other less important data. The
higher the number the higher the priority.
DSCP Voice / VBD / 0 … 46 … 63 Class of service for Voice / VBD / RFC2833
RFC2833 packets.
DSCP Clearmode 0 … 40 … 63 Class of service for clearmode data packets.
DSCP T.38 0 … 40 … 63 Class of service for T.38 packets.
a. If the gateway IP address is changed by the restore process or during the unassign state, the unit will be restarted au-
tomatically.
8.3.3.7 AP: / unit-x, Configuration - Gateway - VLAN
Table 53: AP: / unit-x, Configuration - Gateway - VLAN

VLAN ID 0 … 4’089 Interface VLAN ID.
The VLAN is used for the media and control traf-
fic flows.
Note that the VLAN ID 0 is not a valid value.
Priority CoS 0 (Best Effort) Priority of the interface traffic.
CoS 1 (Background)
CoS 2 (Spare)
CoS 3 (Excellent Effort)
CoS 4 (Controlled Load)
CoS 5 (Video)
CoS 6 (Voice)
CoS 7 (Network Control)

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IPSS4
8.3.3.8 AP: / unit-x, Configuration - Gateway - DNS Table
Add SRV Record (left) and A Record (right) dialogues:
Table 54: AP: / unit-x, Configuration - Gateway - DNS Table

DNS Table SRV Records, Ser- SIP Symbolic name of the desired service.
vice The service is fixed to SIP.
For further information please refer to section
5.5.2.6 "Scenario with manual A record and SRV
record entries" (on page 84).
SRV Records, Pro- UDP Symbolic name of the desired protocol.
tocol The protocol is fixed to UDP.
SRV Records, 0 … 63 characters The domain name for which this record is valid
Name (without starting dot).
E.g. london.voip.com.
The default name is empty.
SRV Records, Pri- 0 … 65’535 The priority of this target host.
ority Lower values mean higher priority.
A target with lower priority (higher value) is only
accessed if the target with higher priority (lower
value) is unavailable.

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Table 54: AP: / unit-x, Configuration - Gateway - DNS Table (continued)

SRV Records, 0 … 65’535 The relative weight for records with the same pri-
Weight ority.
Higher values mean higher probability of being
selected. Traffic is directed to targets propor-
tional to their weight.
SRV Records, Port 0 … 65’535 The UDP port on the target host on which the
service is to be found.
E.g. 5060
SRV Records, Tar- 0 … 63 characters The domain name of the target host
get E.g. proxy-a.london.voip.com.
The default target name is empty.
DNS Table A Records, Name 0 … 63 characters The domain name for which this record is valid
(without starting dot).
E.g. proxy-a.london.voip.com.
The default name is empty.
For further information please refer to section
5.5.2.6 "Scenario with manual A record and SRV
record entries" (on page 84).
A Records, IP <IPv4 address> IP address of the domain.
Address e.g. 172.31.63.181 The default address is 0.0.0.0
8.3.3.9 AP: / unit-x, Configuration - Proxy / Registrar

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Table 55: AP: / unit-x, Configuration - Proxy / Registrar

Proxy Proxy Mode Primary Only Only a primary proxy is configured
Revertive IPSS4 switches to the secondary proxy if the pri-
mary proxy as soon as the primary proxy
becomes available again.
Non Revertive IPSS4 switches to the secondary proxy if the pri-
mary proxy only if the secondary proxy fails.
DNS / RFC3263 No proxy is configured. The proxy is found via
DNS queries.
Primary Proxy Proxy Address <IPv4 address> Outbound primary proxy address as IP or fully
e.g. 172.31.63.181 qualified domain name (FQDN).
<FQDN> A proxy address is required when no DNS server
address is configured. If no proxy is used the
e.g. proxy.london.com
proxy address configuration must be empty and
ther information.
The default proxy address is empty.
Proxy Port 0 … 5060 … 9999 Outbound proxy port number.
Secondary Proxy Proxy Address <IPv4 address> Outbound secondary proxy address as IP or fully
e.g. 172.31.63.181 qualified domain name (FQDN).
<FQDN> A proxy address is required when no DNS server
address is configured. If no proxy is used the
e.g. proxy.london.com
proxy address configuration must be empty and
ther information.
The default proxy address is empty.
Proxy Port 0 … 9999 Outbound proxy port number.
Proxy Availability Enabled  Enable the proxy availability check for this gate-
Check  way.
The proxy availability check is also used to
check the DNS availability in case the proxy
mode is set to “DNS / RFC 3263”.
Method OPTIONS Send an OPTIONS or a REGISTER request to
the proxy to check the availability of the proxy.
REGISTER
Interval 10 … 100 s Enter the interval on which the proxy availability
check shall be performed.
Note that during proxy unavailability the check
interval is 10 s, irrespective of the configuration
value.

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IPSS4
Table 55: AP: / unit-x, Configuration - Proxy / Registrar (continued)

Registrar Registrar Address <IPv4 address> Registrar address configured as IP or fully quali-
e.g. 172.31.63.181 fied domain name (FQDN).
<FQDN> A registrar address is always required even if the
registration mode is set to no registration. Please
e.g. regist.london.com
refer to paragraph 5.5 "Proxy server and regis-
trar server functions" (on page 77) for further
information.
The default registrar address is empty.
Registrar Port 0 … 5060 … 9999 Registrar port number.
Registration Mode No Registration Registration mode with a registrar.
“No Registration” is used if all subscribers are
One-By-One Registration
registered by provisioning in the registrar.
Registration Expira- 1 … 100 h Enter the registration expiration time.
tion Time
Please note:
Changes in the parameters below cause a re-creation of all affected ports,
i.e. the ports are removed and the respective subscriber numbers are de-
registered. The ports are thereafter added again and the respective sub-
scriber numbers are newly registered.
→ Proxy Address,
→ Proxy Port,
→ Registrar Address,
→ Registrar Port,
→ Registration Mode,
→ Registration Expiration Time.
8.3.3.10 AP: / unit-x, Configuration - Digit Map
Add Digit Map table entry dialogue:

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IPSS4
Table 56: AP: / unit-x, Configuration - Digit Map

Digit Map Add… Add a digit map entry.
Up to 16 digitmap entries are configurable.
Remove Delete a selected digit map entry.
URI Schema sip Selection of the URI schema to form the destina-
tion address.
tel
Note that the “sip” URI schema is applied for the
“From:” and “Contact:” header independent of
the URI schema configuration.
Digit Map 0 … 64 characters Digit map string with up to 8 subscriber number-
ing scheme alternatives
Please refer to section 5.4.7 "Digit maps" (on
page 73) for further information.
Domain / Phone 0 … 64 characters Domain address of the telephone service pro-
Context e.g. london.voip.com vider.
Note: If the Domain / Phone Context parameter
is left empty, the IPSS4 unit automatically
selects and uses the actual home domain as it is
configured in the unit configuration or in the con-
figured SIP profile of a subscriber port.
Pre Strip 0 … 16 Number of digits to be removed from the start of
the called number.
Pre Pend 0 … 16 characters Digits to be added at the start of the called num-
ber.
Valid characters are 0-9, * and #.
Timers Start Timer 10 … 40 … 90 s Start timer for the digit collection process.
Short Timer 1 … 5 … 30 s Short timer for the digit collection process.
Long Timer 1 … 10 … 60 s Long timer for the digit collection process.
8.3.3.11 AP: / unit-x, Configuration - CPS

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IPSS4
Table 57: AP: / unit-x, Configuration - CPS

Unit CPS Name <File list> File list of the CPS versions available.
The prefix of a valid CPS file for the IPSS4 unit is
IPSS_.
Preview… Open the preview window of the selected CPS.
This allows you to check all CPS parameters
Please note:
The global configuration parameters are selected as a whole per IPSS4 unit
with the CPS file.
→ KEYMILE produces the CPS file on the basis of a questionnaire which
is filled in by the customer. You can select from all previously down-
loaded CPS files. In general you select the most recent version of a
CPS.
8.3.3.12 AP: / unit-x, Configuration - EQP
Create EQP Group dialogue:
Table 58: AP: / unit-x, Configuration - EQP

Create EQP Open the dialogue for the creation of an equip-
Group ... ment protection (EQP) group.
A unit must be in the unassigned state to be
selectable as a protecting unit.
EQP Group Crea- Group Type 1:1 Type of the EQP group.
tion MO address of the working unit, e.g. /unit-19.
Working Unit /unit-x
In the ECST the working unit MO address is
read-only.
Protecting Unit - List of the MO addresses of all HW compatible
units which are not assigned, e.g. /unit-21.
/unit-y
Default is no unit selected.

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IPSS4
Table 58: AP: / unit-x, Configuration - EQP (continued)

Delete EQP Group Delete an existing protection group.
Traffic on the protecting unit will be interrupted
and services will be re-established on the work-
ing unit.
The protecting unit will become unassigned.
8.3.3.13 AP: / unit-x, Configuration - Port Groups
Create PSTN Port Group (left), Create ISDN-BA Port Group (right) and
Delete Port Group (bottom) dialogues:
Resolve Logical Port dialogues:
Table 59: AP: / unit-x, Configuration - Port Groups

Create PSTN Port Open the dialogue for the creation of a PSTN
Group… portgroup. A portgroup contains a configurable
number of PSTN ports that have to be cross con-
nected to the ports of a PSTN linecard.
PSTN Port Group Group Index 1 … 255 Select the portgroup index number.
Default is the lowest numbered available index
number.
The number of PSTN portgroups is limited to 50
per IPSS4 unit.
Number Of Ports 0 … 48 … 64 Select the number of physical PSTN subscriber
ports to be assigned to the portgroup.

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Table 59: AP: / unit-x, Configuration - Port Groups (continued)

Create ISDN-BA Open the dialogue for the creation of an ISDN-
Port Group… BA portgroup. A portgroup contains a configur-
able number of ISDN-BA ports that have to be
cross connected to the ports of the ISDN-BA
linecard.
ISDN Port Group Group Index 1 … 255 Select the portgroup index number.
Default is the lowest numbered available index
number.
The number of ISDN-BA portgroups is limited to
20 per IPSS4 unit.
Number Of Ports 0 … 16 Select the number of physical ISDN-BA sub-
scriber ports to be assigned to the portgroup.
Delete Port Open the dialogue for the deletion of a PSTN or
Group… ISDN-BA portgroup, i.e. delete all assigned
physical PSTN or ISDN-BA subscriber ports
from the gateway and the corresponding port-
group from the AP tree.
Port Group Port Group portgroup-1 … portgroup- Select a single portgroup to be deleted or all
255 portgroups.
All
Resolve Logical Open the dialogue for the logical port resolution.
Port… By entering the subscriber number, the unit num-
ber, portgroup number and port number where
the subscriber number is assigned to, is dis-
played.
The port must be operational to be able to be
resolved.
Note that only subscriber numbers assigned to
the gateway unit where the command is issued,
can be resolved.
Logical Port Gateway/Interface 0 … 63 characters The gateway or interface ID is only required
ID when accessing the network element with
UNEM.
This parameter can remain empty when access-
ing the network element with ECST.
Subscriber ID 0 … 63 characters Local subscriber number as configured in the
port configuration or global subscriber number,
starting with “0” or with the “+” sign.
8.3.4 AP: / unit-x, Fault Management
For the a description of the general aspects of the

− “Fault Management - Status”, and
− “Fault Management - Configuration”
2510”. The following table lists the fault causes of the current AP.

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Table 60: AP: / unit-x, Fault Management

ID Fault Cause Event Type Traffic Default Description
Affecting Severity
SWM Software Mismatch Equipment  Minor The running ESW does not match the
Alarm assigned ESW.
SSWNA Scheduled Software Equipment  Minor The ESW that is scheduled for installa-
Not Available Alarm tion is not available on the unit. Make
sure that the ESW is downloaded to the
unit.
RSWNA Running Software Not Equipment  Major The running ESW has been deleted from
Available Alarm the unit flash memory.
The unit is still able to provide its service,
but a new ESW is required to be down-
loaded to the unit and assigned.
SWIN Software Incompatible Equipment  Major The running ESW is not compatible with
With Network Element Alarm the version required by the NE type or
version.
PRC PBUS Resource Con- Processing  Major The PBUS access circuit of another TDM
flict Error Alarm unit in the subrack is defective.
Remove or unassign the other TDM units
in the subrack one by one until the alarm
is cleared. Replace the defective unit.
EQM Equipment Malfunction Equipment  Critical The IPSS4 controller detects any anom-
Alarm alies on the unit, e.g. a voltage is miss-
ing, a chip does not respond, etc.
PRAF Proxy Availability Fail- Communica-  Major The proxy did not respond to the
ure tion Alarm OPTIONS or REGISTER request mes-
sage sent by the gateway.
DNSAF Domain Name Server Communica-  Major The preferred and the alternate DNS
Availability Failure tion Alarm servers give no response.
SNRF Subscriber Number Communica-  Minor An enabled subscriber has not been reg-
Registration Failure tion Alarm istered with the registrar server.
The alarm is cleared when all enabled
subscribers have been successfully reg-
istered.
G729DIS G729 Disabled in CPS Communica-  Minor The G.729 AB codec is enabled in the
tion Alarm PSTN or ISDN-BA port profile but disa-
bled in the CPS. To be usable the G.729
AB codec must be enabled in the profile
and in the CPS.
MACE Max Active Calls Communica-  Minor The number of active calls exceeds the
Exceed tion Alarm limit (typically 200 active calls).
The alarm is cleared as soon as the
number of active calls is lower than 90%
of the limit (typically 180 active calls).
No new calls can be setup as long as the
number of active calls exceeds the limit.
If the alarm persists please contact a
KEYMILE representative.
ICPS Incomplete CPS Ver- Processing  Major The configured CPS is not complete or
sion Error Alarm outdated, i.e. from a previous release.
Contact KEYMILE to get a valid CPS.

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IPSS4
Table 60: AP: / unit-x, Fault Management (continued)

Affecting Severity
RANS Registrar Address Not Processing  Major There is no registrar address specified at
Specified Error Alarm startup of the IPSS4 unit.
You have to configure a registrar
address to clear the alarm.
OHTE Off-Hook Timeout Processing  Warning At least one PSTN subscriber has been
Exceed Error Alarm off-hook for a certain time without being
connected to another subscriber. The
affected subscriber gets the howler tone
played.
The alarm is cleared when the sub-
scriber goes on-hook again or becomes
blocked.
The off-hook timer is CPS controlled.
Note that when a subscriber goes off-
hook the alarm is set when the start
timer elapses and subsequently the off-
hook timer elapses.
HWIC Hardware Incompatible Equipment  Major The plugged HW is not compatible with
With Configuration Alarm the unit configuration HW stored in the
database. You may need to change the
HW or re-create the configuration for the
unit.
SWIC Software Incompatible Equipment  Major The ESW running on the unit is not com-
With Configuration Alarm patible with the unit configuration stored
in the database. You may need to
upgrade, or downgrade the ESW, or re-
create the configuration with the actually
running ESW.
GSW General Software Equipment  Major An ESW internal error has been detected
Alarm Alarm that might inhibit the ESW from running
correctly.
MFA Maintenance Function Communica-  Warning A maintenance function has been acti-
Active tion Alarm vated by the operator.
UNAV Unit Not Available Equipment  Critical The unit that is configured is either not
Alarm plugged or not recognized due to a fail-
ure.
NSW No Application Soft- Equipment  Major There is no application ESW installed on
ware Alarm the unit, or the application ESW has not
yet finished its boot process.
UNAS Unit Not Assigned Equipment  Warning The unit is not assigned and can not be
Alarm configured. To assign the unit, execute
the “Assign” command in the “Main”
function of the unit.
UIC Unit Incompatible Equipment  Major The inserted unit is not compatible with
Alarm the assigned unit.
PWRSVE Battery Power Saving Equipment  Critical Power saving is active on the unit, i.e. it
Alarm is kept in the “reset” state during battery
power backup.
EQPUNV EQP Unit Not Available Equipment  Major The standby unit is not available, i.e. not
Alarm plugged in or has an equipment failure.
This alarm is available on the working
and the protecting unit of an EQP group.

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IPSS4
Table 60: AP: / unit-x, Fault Management (continued)

Affecting Severity
EQPHWI EQP Hardware Incom- Equipment  Major The actual hardware is not compatible
patible With Configura- Alarm with the hardware of the protecting unit.
tion This alarm is available on the working
unit of an EQP group only.
EQPSWI EQP Software Incom- Equipment  Major The running software (ESW) is not com-
patible With Configura- Alarm patible with the software on the protect-
tion ing unit.
EQPNPR EQP Group Not Pro- Equipment  Major The working or the protecting unit is not
tected Alarm available, i.e. not plugged in or has an
equipment failure.
EQPWUI EQP Working Unit Iso- Equipment  Major The working unit has been isolated with
lated Alarm the “Isolate Unit” status command.
No protection is available in this state.
EQPUNA EQP Working Unit Not Equipment  Major An automatic protection switch or a
Active Alarm “manual switch” or a “forced switch” to
the protecting unit has been done.
Please note:
Automatic, manual and forced protection switching is available from the
working to the protecting unit and vice versa.
Please refer to section 5.7 "Equipment protection (EQP)" (on page 88).
8.3.5 AP: / unit-x, Performance Management
For the a description of the general aspects of the performance management

(PM) functions, please refer to [302] User Guide “MileGate 2310/2510”.
The PM parameters are presented in different groups. The following counter
groups are available for the IPSS4 unit:
• “Call Statistics” group, see section 8.3.5.1 "AP: / unit-x, Performance
Management - Call Statistics" (on page 163),
• “SIP Statistics” group, see section 8.3.5.2 "AP: / unit-x, Performance
Management - SIP Statistics" (on page 163),
• “RTP MIB” group, see section 8.3.5.3 "AP: / unit-x, Performance Manage-
ment - RTP MIB" (on page 163),
In each of the counter groups, the following counter intervals are available:
• User Counter
• History 15min
• History 24h

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IPSS4
8.3.5.1 AP: / unit-x, Performance Management - Call Statistics
Table 61: PM group: Call Statistics

PM parameter Description
Total Calls Total calls, PSTN and ISDN
Short Calls Calls shorter than 20 seconds (duration of the RTP
packet transmission).
The maximum time for a short call is pre-provisioned
in the CPS.
Voice Calls Total number of calls that were not fax or modem
calls
Fax Calls Total number of fax calls
Modem Calls Total number of modem calls
G711 PT Number of times a payload type 8 was applied a
G729 PT Number of times a payload type 18 was applied a
T38 PT Number of times a payload type T38 was applied
a. Note that more than one payload type can be counted for a call, e.g. first G.729 and
then G.711 for a modem call.
8.3.5.2 AP: / unit-x, Performance Management - SIP Statistics
Table 62: PM group: SIP Statistics

Generated Number of calls generated, i.e. initiated by the local
end
Gen Established Number of calls generated and established
Gen Rejected Number of calls generated and rejected
Received Number of calls received, i.e. initiated by the remote
end
Recv Established Number of calls received and established
Recv Rejected Number of calls received and rejected
Terminated Number of calls terminated by either end
Gen Transactions Number of SIP transactions sent by the local end.
Only two-way handshakes not related to session
establishment, e.g. REGISTER-OK
Recv Transactions Number of transactions received by the local end.
Only two-way handshakes not related to session
establishment.
8.3.5.3 AP: / unit-x, Performance Management - RTP MIB
Table 63: PM group: RTP MIB

In Packets Count of RTP packets received by this RTP host
since start time
Out Packets Count of RTP packets sent by this RTP host since
start time

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IPSS4
Table 63: PM group: RTP MIB (continued)

In RTCP Packets Count of RTCP packets received
Out RTCP Packets Count of RTCP packets sent
8.3.6 AP: / unit-x, Status
8.3.6.1 AP: / unit-x, Status - Proxy
Table 64: AP: / unit-x, Status - Proxy

Availability, Proxy/ Profile <Profile Name> Profile the proxy belongs to.
Registrar List
none
Address <IPv4 address> Proxy IP address or FQDN.
<FQDN>
Port 1 … 65’535 Proxy port.
Mode Primary Only Configured proxy mode.
Revertive
Non Revertive
DNS / RFC3263
Active No Proxy is not active (standby or not configured).
Yes Proxy is active.
Available No Provides availability information about the out-
bound proxy by checking whether OPTIONS or
Yes
REGISTER requests are replied.
Not Checked Status “Not Checked” is displayed when the
proxy availability check is disabled.

207
IPSS4
8.3.6.2 AP: / unit-x, Status - Gateway
Ping Request (left) and Results (right) dialogues:
Please note:
DNS query answers are cached in the local resolver only when enabled by
the CPS. The CPS setting has no influence on the manual DNS entries pro-
visioned via the element manager.

207
IPSS4
Table 65: AP: / unit-x, Status - Gateway

DNS Cache SRV Records, 0 … 63 characters Symbolic name of the service, i.e. SIP.
_Service._Protocol.N Symbolic name of the protocol, i.e. UDP.
ame Domain name for which this record is valid.
SRV Records, TTL 0 … 264-1 The time to live of this record entry in seconds.
The TTL of manual entries shows “0”, meaning
that these records are valid indefinitely.
SRV Records, Prior- 0 … 65’535 The priority of the target host.
ity Lower values mean higher priority.
SRV Records, 0 … 65’535 The relative weight of this record.
Weight Higher values mean higher probability of being
selected.
SRV Records, Port 0 … 65’535 The UDP port on the target host on which the
service is to be found.
SRV Records, Target 0 … 63 characters The domain name of the target host.
A Records, Name 0 … 63 characters The domain name for which this record is valid.
A Records, TTL 0… 264-1 The time to live of this record entry in seconds.
The TTL of manual entries shows “0”, meaning
that these records are valid indefinitely.
A Records, IP <IPv4 address> IP address of the domain.
Address
PenaltyBox Destinations, IP <IP Address> IP address of the unreachable target.
Address
Destinations, Port 0 … 65’535 Port number of the penalty box entry.
Destinations, TTL 0… 264-1 Time to live in seconds of the penalty box entry.
Ping… Open the IP ping request window.
IP Ping Request Destination Address <IPv4 address> IP address of the equipment to be pinged.
e.g. 172.31.63.181 The default destination address is empty.
Number Of Request 0…3…5 Enter the number of ping requests.
Size Of Packets 0 … 32 … 3999 Bytes Enter the size of the ping packets.
IP Ping Response Number Of Echo 0…5 Number of sent ping packets.
Requests
Number Of Echo 0…5 Number of received ping packets.
Responses
8.3.6.3 AP: / unit-x, Status - Registration

207
IPSS4
Table 66: AP: / unit-x, Status - Registration

Registration, Unreg- Subscriber Number 0 … 20 characters Unregistered subscriber number.
istered Subscriber Transient state after (re-)configuration of the
Registration Status - - -
List port.
Unregistered The registering process has not started yet.
Registering Registering in progress. The register request has
been sent, awaiting a response.
Registering With Cred Registering in progress. A 401 (Unauthorized) or
407 (Proxy Authentication Required) response
has been received after the first register request.
A new register request with credentials is sent.
Reregistering Registration has expired. A new register request
has been sent, awaiting a response.
DeReg In Registering A port has been (re-)configured during the regis-
tering process.
De-Registering De-Registering in progress. The register request
with 0 expiration time has been sent, awaiting a
response.
Address /portgroup-y/port-z Portgroup and port address of the unregistered
subscriber number.
8.3.6.4 AP: / unit-x, Status - Ongoing Calls
Table 67: AP: / unit-x, Status - Ongoing Calls

Ongoing Calls Number Of Calls, 0 … 65’535 Number of ongoing PSTN calls on the unit.
PSTN
Number Of Calls, 0 … 65’535 Number of ongoing ISDN-BA calls on the unit.
ISDN-BA

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IPSS4
Table 67: AP: / unit-x, Status - Ongoing Calls (continued)

PSTN, Local Num- 0 … 30 characters Calling number of a PSTN call.
ber
PSTN, Remote 0 … 30 characters Called number of a PSTN call.
Number
PSTN, SIP Call 0 … 40 characters Value of the internal state machine.
Control
PSTN, PSTN Call 0 … 40 characters Value of the internal state machine.
Control
PSTN, PSTN Sup- 0 … 40 characters Value of the internal state machine.
plementary Service
PSTN, Codec 0 … 40 characters Used codec for the call.
ISDN-BA, Local 0 … 30 characters Calling number of an ISDN-BA call.
Number
ISDN-BA, Remote 0 … 30 characters Called number of an ISDN-BA call.
Number
ISDN-BA, SIP Call 0 … 40 characters Value of the internal state machine.
Control
ISDN-BA, ISDN-BA 0 … 40 characters Value of the internal state machine.
Call Control
ISDN-BA, Codec 0 … 40 characters Used codec for the call.
Warnings PSTN Off-Hook, 0 … 30 characters Local number of the PSTN port exceeding the
Local Number off-hook time without establishing a call.
Refer also to the OHTE alarm.
PSTN Off-Hook, 0 … 63 characters Portgroup number and port number of the PSTN
PSTN Port port exceeding the off-hook time without estab-
lishing a call.
Please note:
A gateway internal call shows two entries in the ongoing calls tables.
8.3.6.5 AP: / unit-x, Status - EQP

207
IPSS4
Table 68: AP: / unit-x, Status - EQP

EQP Group Manual Switch-Over  Indication that the standby unit, i.e. the working
Allowed or the protecting unit is operational and can take
over the service.
 Indication that the standby unit, i.e. the working
or the protecting unit is not operational due to a
failure or to isolation or that the database has not
been saved. A manual protection switching with
the “Manual Switch-Over” command is not possi-
ble.
EQP Group - Units Unit <MO address> MO address of the unit belonging to the EQP
Status group.
EQP Unit Mode Undefined Mode Mode of a unit in the EQP group.
Working Unit
Protecting Unit
Active  The unit is active (operational).
 The unit is standby (not operational).
Failure  The unit has detected failure(s).

Substituted  The working unit has been substituted by the
 protecting unit.
Isolated  The working unit has been isolated with the “Iso-
 late Unit” command.
HW Compatible  The working unit is hardware compatible with the

 protecting unit in the EQP group.
Please refer to section 5.7.2 "EQP prerequisites"
(on page 88).
SW Compatible  The working unit is embedded software compati-
 ble with the protecting unit in the EQP group.
Please refer to section 5.7.2 "EQP prerequisites"
(on page 88).
DB Saved  The configuration of the unit has been saved.

Manual Switch-Over Manual switch of the unit in the EQP group.
A switch over is performed only when the other
unit is error free and is not isolated and the data-
base has been saved.
Refer to the “Manual Switch-Over Allowed” prop-
erty above.
Note: The traffic will be interrupted for about
20 s.
Forced Switch-Over Forced switch of the unit in the EQP group.
A switch over is performed independent of the
error state of the other unit., but the unit must not
be isolated.
Note: The traffic will be interrupted for about
20 s.
Note: Traffic will remain interrupted if the active
unit is not operational.

207
IPSS4
Table 68: AP: / unit-x, Status - EQP (continued)

Isolate Unit Isolate the working unit from the EQP group.
An isolated working unit will not perform any pro-
tection switch action. The protection switching
state of the unit is frozen.
Join Unit Join the isolated working unit to the EQP group,
i.e. remove the isolation.

207
IPSS4
8.4 AP: / unit-x / control
8.4.1 AP: / unit-x / control, Overview

− “Overview - Alarms”
management function, please refer to [302] User Guide “MileGate 2310/
2510”.
8.4.2 AP: / unit-x / control, Main

− “Main - General”
2510”.

207
IPSS4
8.5 AP: / unit-x / media
8.5.1 AP: / unit-x / media, Overview

2510”.
8.5.2 AP: / unit-x / media, Main

2510”.

207
IPSS4
8.6 AP: / unit-x / huntgroups
Please note:
8.6.1 AP: / unit-x / huntgroups, Overview

2510”.
8.6.2 AP: / unit-x / huntgroups, Main

2510”.
8.6.3 AP: / unit-x / huntgroups, Configuration
8.6.3.1 AP: / unit-x / huntgroups, Configuration - Create Group
Create Group (left) dialogue:
Table 69: AP: / unit-x / huntgroups, Configuration - Create Group

Create Group … Open the dialogue to add a new hunting group.
Please refer to section 5.6 "Direct dialling in and
trunk hunting" (on page 86) for further details.

207
IPSS4
Table 69: AP: / unit-x / huntgroups, Configuration - Create Group (continued)

Create Hunt Group Port Type ISDN-BA DDI Numbers Hunting group with individually configurable DDI
numbers.
ISDN-BA DDI Ranges Hunting group with DDI numbers configurable as
ranges.
Group Number 1 … 999 Identification number of the hunting group.
Default is the lowest available number.
In practice the number of huntgroups is limited to
304 according to the number of available ISDN-
BA ports in the subrack.
8.6.3.2 AP: / unit-x / huntgroups, Configuration - Delete Group
Create Group (left) and Delete Group (right) dialogues:
Table 70: AP: / unit-x / huntgroups, Configuration - Create Group

Delete Group … Open the dialogue to delete an existing hunting
group.
Hunt Group Hunt Group group-a Select one hunting group or all hunting groups to
be deleted.
all
Default is the lowest numbered hunting group.

207
IPSS4
8.7 AP: / unit-x / huntgroups / group-a (ISDN-BA)
8.7.1 AP: / unit-x / huntgroups / group-a, Overview

2510”.
8.7.2 AP: / unit-x / huntgroups / group-a, Main

2510”.
8.7.3 AP: / unit-x / huntgroups / group-a, Configuration
8.7.3.1 AP: / unit-x / huntgroups / group-a, Configuration - Port
Add Port (left) and Delete Port (right) dialogues:
Table 71: AP: / unit-x / huntgroups / group-a, Configuration - Port

Add Port … Open the dialogue to add a new port to the hunt-
ing group.
Please refer to section 5.6 "Direct dialling in and
trunk hunting" (on page 86) for further details.
Click the Apply or Refresh button after adding or
deleting a port.
Only ports available in an existing portgroup
which are not enabled can be added to a hunt-
ing group.

207
IPSS4
Table 71: AP: / unit-x / huntgroups / group-a, Configuration - Port (continued)

Hunt Group Port Name portgroup-y/port-z Available port to be added to the hunting groups.
Default is the lowest numbered port.
Delete Port … Open the dialogue to delete a port from a hunt-
ing group.
Hunt Group Port Name portgroup-y/port-z Select one port or all ports to be deleted from the
hunting group.
all
Default is the lowest numbered port.
Associated Ports, Port Name portgroup-y/port-z List of all ports which are member of the hunting
Port Names group.
8.7.3.2 AP: / unit-x / huntgroups / group-a: DDI Numbers, Configuration - Settings
Set DDI Numbers dialogue:
Add Subscriber Identifications table dialogue:

207
IPSS4
Add Screening Table entry dialogue:
Table 72: AP: / unit-x / huntgroups / group-a: DDI Numbers, Configuration - Settings
Set DDI Numbers… Open the dialogue to enter the ISDN-BA sub-
scriber numbers as a range. Only numbers that
are not already in use on IPSS4 can be entered.
To add authorisation parameters please use the
“Add…” command described below.
Note: This deletes all previous entries.
After entering a new number range press also
the Refresh button in the configuration window
to display the new entries.
ISDN DDI Number Start DDI Number 0 … 20 characters Enter the first DDI number.
Range Enter the number of DDI numbers to be added.
Number Range 1 … 100
The numbers added are incremented by 1 start-
ing with the “Start DDI Number”.
Default Number Default Number 0 … 20 characters Displays the default number. The default number
is the first subscriber number in the subscriber
identifications list.
ISDN Hunt Group Enable  Enable the ISDN-BA hunting group.

ISDN Hunt Group, Add… Add a single subscriber identification to the sub-
Subscriber Identifi- scriber identifications list.
cations To modify a single subscriber identification
select the identification to be modified by clicking
on it.
Remove Delete a single subscriber identification. Select
the number to be removed by clicking on it.
Subscriber Number 0 … 20 characters Subscriber number belonging to the hunting
group.
The list contains in maximum 100 entries.
The default subscriber number is empty.
Authorisation User 0 … 63 characters The authorisation user name is used for the
Name HTTP digest authentication.
Typically this is the subscriber number including
the area and country codes.
The default authorisation user name is empty.
This field can be left empty if it is not required.

207
IPSS4
Table 72: AP: / unit-x / huntgroups / group-a: DDI Numbers, Configuration - Settings (continued)
Authorisation Pass- 0 … 63 characters The authorisation password is used for the
word HTTP digest authentication.
The default authorisation password is empty.
Display Name 0 … 63 characters Display names are used in SIP URIs and in the
P-…-Identity headers.
The display name is optional and can be left
empty if not required.
In the CPS it is configurable if display names or
subscriber numbers are used for the CLIP ser-
vice.
The default display name is empty.
Privacy None Privacy header type to be used.
Id The Privacy header is only inserted if “Privacy” is
set to “Id” and if the asserted identity mode is
configured to “Asserted” or “Preferred”.
and privacy" (on page 53).
ISDN Hunt Group DDI Prefix Length 0 … 20 Number of DDI number digits that are not for-
warded towards the subscriber.
This excludes the country code and the area
code.
Refer also to section 5.6 "Direct dialling in and
trunk hunting" (on page 86).
Register As Global  Expand the contact address field with the coun-
 try and area code for registration, i.e. you enter
the subscriber number without country and area
code.
Register Default  Register the default number only or register all
Number only  numbers.
Layer1 Perma-  With the permanently activated ISDN-BA layer 1,

nently Activated  the digital section and the S-Bus including the
terminal equipment are activated. This allows a
shorter set-up time for a call.
S-Bus activation is only possible if a TE is con-
nected to it.
If layer 1 is not activated permanently, it will be
activated per call.
SIP Profile <List of downloaded SIP Select the appropriate profile from the drop down
profiles> list.
none Select “none” to apply the corresponding unit
configuration parameters of the AP: /unit-x, Con-
figuration - Gateway - SIP.
Proxy / Registrar <List of downloaded Proxy Select the appropriate profile from the drop down
Profile / Registrar profiles> list.
figuration - Proxy / Registrar.
Codec / SDP Profile <List of downloaded Codec Select the appropriate profile from the drop down
/ SDP profiles> list.
figuration - Gateway - Media - Codec / SDP.

207
IPSS4
Table 72: AP: / unit-x / huntgroups / group-a: DDI Numbers, Configuration - Settings (continued)
ISDN-BA Profile <List of downloaded ISDN- Select the appropriate profile from the drop down
BA profiles> list.
figuration - Gateway - Supplementary Services -
Available Services.
Preview This command opens a window displaying the
parameters of the actually selected profile. The
contents are identical to those described in the
section 8.2 "Profiles" (on page 131).
Special Arrange- Enable SA  Enable the “CLIP no screening” feature.
ment  With the enabled special arrangement the IPSS4
forwards the configured valid user provided num-
bers of the screening table to the SIP headers.
The valid user provided numbers need not to be
a configured DDI number.
With the disabled special arrangement the
IPSS4 does only forward a number which is a
configured DDI number.
Please refer also to section 5.4.3.5 "Calling line
ID presentation (CLIP)" (on page 64).
Special Arrange- Add… Add a valid user number to the screening table.
ment, Screening You can add up to 16 user numbers to the table.
Table
Remove Delete a user number from the screening table.
Select the number to be removed by clicking on
it.
Valid User Pro- 0 … 20 characters The user number is entered as digit map string
vided Numbers with the following syntax:
- Allowed characters: 0 - 9.
- x = wildcard, replacing a symbol in the range
0 - 9.
- . = zero or more repetitions of the character or
wildcard that precedes it. A “.” must be placed
at the end of digit map string.
Please note:
The hotline supplementary service is not supported by a hunting group port.
→ The hotline parameter setting in the ISDN-BA profile is ignored.
Please note:
ETSI defines:
A special arrangement is an agreement between a customer and a public
network operator whereby customer supplied connected party ISDN num-
bers are not screened by the public ISDN.
Please note:
ETSI defines:
Screening is a process whereby the network checks that a user provided
information is acceptable to the network.

207
IPSS4
8.7.3.3 AP: / unit-x / huntgroups / group-a: DDI Ranges, Configuration - Settings
Add DDI Ranges dialogue:
Add Screening Table entry dialogue:
Table 73: AP: / unit-x / huntgroups / group-a: DDI Ranges, Configuration - Settings
ISDN Hunt Group Enable  Enable the ISDN-BA hunting group.


207
IPSS4
Table 73: AP: / unit-x / huntgroups / group-a: DDI Ranges, Configuration - Settings (continued)
DDI Prefix Length 0 … 20 Number of DDI number digits that are not for-
warded towards the subscriber.
This excludes the country code and the area
code.
Refer also to section 5.6 "Direct dialling in and
trunk hunting" (on page 86).
Default Number 0 … 20 characters Default number to be used.
The default number is empty.
vice.
Id The Privacy header is only inserted if “Privacy” is
ISDN Hunt Group, Add… Add a hunting group number range.
DDI Ranges Up to 10 number ranges can be added to a hunt-
ing group.
To modify a hunting group number range select
the first or last number to be modified by clicking
on it.
Remove Delete a hunting group number range. Select the
number range to be removed by clicking on it.
First DDI Number 0 … 9 characters Lowest subscriber number belonging to the hunt-
ing group number range.
Last DDI Number 0 … 9 characters Highest subscriber number belonging to the
hunting group number range.
The number of used digits must be the same as
for the first DDI number.
The maximum number range comprises 1000
subscriber numbers.
ISDN Hunt Group Register As Global  Expand the contact address field with the coun-
try and area code for registration, i.e. you enter

code.

207
IPSS4
Layer1 Perma-  With the permanently activated ISDN-BA layer 1,
nected to it.
activated per call.
profiles> list.
BA profiles> list.
Available Services.
Special Arrange- Enable SA  Enable the “CLIP no screening” feature.
ment  With the enabled special arrangement the IPSS4
forwards the configured valid user provided num-
bers of the screening table to the SIP headers.
The valid user provided numbers need not to be
a configured DDI number.
With the disabled special arrangement the
IPSS4 does only forward a number which is a
configured DDI number.
Please refer also to section 5.4.3.5 "Calling line
ID presentation (CLIP)" (on page 64).
Special Arrange- Add… Add a valid user number to the screening table.
ment, Screening You can add up to 16 user numbers to the table.
Table
Remove Delete a user number from the screening table.
Select the number to be removed by clicking on
it.

207
IPSS4
Valid User Pro- 0 … 20 characters The user number is entered as digit map string
vided Numbers with the following syntax:
- Allowed characters: 0 - 9.
- x = wildcard, replacing a symbol in the range
0 - 9.
- . = zero or more repetitions of the character or
wildcard that precedes it. A “.” must be placed
at the end of digit map string.
Please note:
The hotline supplementary service is not supported by a hunting group port.
→ The hotline parameter setting in the ISDN-BA profile is ignored.
Please note:
ETSI defines:
A special arrangement is an agreement between a customer and a public
network operator whereby customer supplied connected party ISDN num-
bers are not screened by the public ISDN.
Please note:
ETSI defines:
Screening is a process whereby the network checks that a user provided
information is acceptable to the network.

207
IPSS4
8.8 AP: / unit-x / portgroup-y (PSTN) or (ISDN-BA)
Please note:
8.8.1 AP: / unit-x / portgroup-y, Overview

− “Overview - Alarms”, and
− “Overview - Cross Connections”
2510”.
8.8.2 AP: / unit-x / portgroup-y, Main

2510”.
8.8.3 AP: / unit-x / portgroup-y, Configuration
8.8.3.1 AP: / unit-x / portgroup-y, Configuration - General
Create Port (left) and Delete Port (right) dialogues of a PSTN portgroup:
Create Port (left) and Delete Port (right) dialogues of an ISDN-BA portgroup:
Table 74: AP: / unit-x / portgroup-y, Configuration - General

Create Port … Open the dialogue for the creation of a PSTN or
ISDN-BA port and adding it to an existing PSTN
or ISDN-BA portgroup.

207
IPSS4
Table 74: AP: / unit-x / portgroup-y, Configuration - General (continued)

PSTN Port Port Index 0 … 64 Subscriber port number in the PSTN portgroup.
Default is the lowest available index number.
The port index 0 automatically assigns the low-
est available index number in the range 1 … 64.
ISDN-BA Port Port Index 0 … 16 Subscriber port number in the ISDN-BA port-
group.
Default is the lowest available index number.
The port index 0 automatically assigns the low-
est available index number in the range 1 … 16.
Delete Port … Open the dialogue for the deletion of a PSTN or
ISDN-BA port. The port is removed from the port
group and the cross connection assigned to the
port is deleted.
Port Port port-1 … port-64 Remove a single or all subscriber ports from the
portgroup.
All
Note that in an ISDN-BA portgroup the highest
port number is port-16.

207
IPSS4
8.9 AP: / unit-x / portgroup-y / port-z (PSTN)
8.9.1 AP: / unit-x / portgroup-y / port-z, Overview

− “Overview - Alarms”,
− “Overview - Cross Connections”, and
− “Overview - CTP”
2510”.
8.9.2 AP: / unit-x / portgroup-y / port-z, Main (PSTN)

2510”.
8.9.3 AP: / unit-x / portgroup-y / port-z: Configuration (PSTN)
8.9.3.1 AP: / unit-x / portgroup-y / port-z: Configuration - Port PSTN

207
IPSS4
Table 75: AP: / unit-x / portgroup-y / port-z: Configuration - Port PSTN

PSTN Port Enable  Enable the PSTN user port.

PSTN Port, Sub- Add… Add a single subscriber identification (MSN num-
scriber Identifica- ber) to the subscriber identifications list. Up to 3
tions PSTN subscriber identifications can be config-
ured.
Subscriber Number 0 … 20 characters Subscriber number of the PSTN user port.
vice.
The Privacy header is only inserted if “Privacy” is
Id
PSTN Port Register As Global  Expand the contact address field with the coun-

code.

207
IPSS4
Table 75: AP: / unit-x / portgroup-y / port-z: Configuration - Port PSTN (continued)
Payphone  With payphone set to true, the IPSS4 applies a
 polarity reversal at the begin of a call and applies
the normal polarity at the end of a call.
This can be used by payphones to control the
call charging.
profiles> list.
PSTN Profile <List of downloaded PSTN Select the appropriate profile from the drop down
profiles> list.
Available Services.
Enterprise Profile <List of downloaded enter- Select the appropriate profile from the drop down
prise profiles> list.
Available Services.

207
IPSS4
8.9.3.2 AP: / unit-x / portgroup-y / port-z, Configuration - CTP
Table 76: AP: / unit-x / portgroup-y / port-z, Configuration - CTP

CTP Configuration Layer Rate P0_nc The layer rate of the connection termination
point is P0_nc, i.e. n x 64 kbit/s.
The layer rate property of a PSTN user port is
fixed to P0_nc.
n 0 … 2 characters Number of timeslots in case of P0_nc. The pos-
sible range is from 1 to 32.
The number of timeslots of a PSTN user port is
fixed to 1.
Timeslot(s) 0 … 64 characters Used timeslots in a structured P12 signal in case
of P0_nc, e.g. 1 … 31.
The timeslot(s) property of a PSTN user port is
empty.
Connected to CTPs Connection Index 0 … 65‘535 Connection index.
Local Role z-End Local role for this connection.
A connection is always established between an
a-End Working
a-end and the z-end.
a-End Protecting For further information refer to [314] User Guide
“TDM Services and Cross Connections”.
Remote CTP MO Address Managed object address of the CTP (connec-
tion termination point) where the PSTN user port
is connected to, e.g. /unit-1/port-1.
Remote Role z-End Remote role for this connection.
a-End Working
Directionality Bidirectional Directionality of the connection.
Unidirectional

207
IPSS4
Table 76: AP: / unit-x / portgroup-y / port-z, Configuration - CTP (continued)

z-End Configuration Revertive Protection  Enable revertive protection switching.
Switching  The z-End will preferably select the working a-
End.
Note that the non-revertive protection switching
will be supported in a future release.
CAS AIS Supervi-  Use CAS AIS as protection switching criterion.
sion 
Switch-Over Log-  Enable the logging of the protection switch-over
ging  events.
8.9.4 AP: / unit-x / portgroup-y / port-z: Status (PSTN)
8.9.4.1 AP: / unit-x / portgroup-y / port-z, Status - Port Status
Table 77: AP: / unit-x / portgroup-y / port-z, Status - Port Status

User Port Status Admin State Locked User port locking state / administrative state
(X.731), controlled by the linecard.
Shutting Down
Unlocked
Oper State Enabled User port locking state / operational state
(X.731).
Disabled
Please note:
8.9.4.2 AP: / unit-x / portgroup-y / port-z, Status - Supplementary Services

207
IPSS4
Table 78: AP: / unit-x / portgroup-y / port-z, Status - Supplementary Services

Call Waiting Activated  User port status of the call waiting service.

Hotline Subscriber Number 0 … 20 characters PSTN subscriber number of the selected port.
Hotline Number 0 … 20 characters Configured hotline number.
- No hotline number is configured or the supple-
mentary service is not activated.
Activated  Hotline supplementary service of the PSTN sub-
 scriber number.
8.9.4.3 AP: / unit-x / portgroup-y / port-z, Status - Registration
Table 79: AP: / unit-x / portgroup-y / port-z, Status - Registration

Registration, Sub- Subscriber Number 0 … 20 characters PSTN subscriber number.
scriber List Transient state after (re-)configuration of the
Registration Status ---
port.
Registered The subscriber number is registered.
tering process.
response.

207
IPSS4
8.9.4.4 AP: / unit-x / portgroup-y / port-z, Status - Ongoing Calls
Table 80: AP: / unit-x / portgroup-y / port-z, Status - Ongoing Calls

Ongoing Calls PSTN, Local Num- 0 … 30 characters Calling number of a PSTN call.
OngoingCallsList ber
PSTN, Remote 0 … 30 characters Called number of a PSTN call.
Number
PSTN, SIP Call 0 … 40 characters Value of the internal state machine.
Control
PSTN, PSTN Call 0 … 40 characters Value of the internal state machine.
Control
PSTN, PSTN Sup- 0 … 40 characters Value of the internal state machine.
plementary Service
PSTN, Codec 0 … 40 characters Used codec for the call.
8.9.4.5 AP: / unit-x / portgroup-y / port-z: Status - CTP

207
IPSS4
Table 81: AP: / unit-x / portgroup-y / port-z: Status - CTP

Trail Status Working Trail, <MO Address> Managed object address of the CTP (connec-
Remote CTP tion termination point) where the P0 signal is
connected to, e.g. /unit-1/port-32.
Working Trail, Sta- OK No failure on the received signal.
tus Signal Fail status on the received signal.
SF
CAS AIS CAS AIS status on the received signal.
This status is only available if CAS AIS supervi-
sion is enabled.
SD Signal Degraded status on the received signal.
Not Available The status of the received signal is not available,
e.g. when the CTPs role is a-end in a unidirec-
tional connection.
Protecting Trail, <MO Address> Managed object address of the CTP (connec-
Remote CTP tion termination point) where the P0 signal is
Protecting Trail, OK No failure on the received signal.
Status Signal Fail status on the received signal.
SF
CAS AIS CAS AIS status on the received signal.
This status is only available if CAS AIS supervi-
sion is enabled.
tional connection.
Active Trail Working The trail from the a-end working remote CTP has
been selected.
Protecting The trail from the a-end protecting remote CTP
has been selected.
Not Available There is no active trail.
External Request Request Release Automatic trail selection.
Force Working Force the selector to use the trail from the a-end
working remote CTP.
Force Protecting Force the selector to use the trail from the a-end
protecting remote CTP.
Manual Working Prefer the trail from the a-end working remote
CTP. Use this trail only if the fault status is not
worse than the fault status of the protecting trail.
Manual Protecting Prefer the trail from the a-end protecting remote
worse than the fault status of the working trail.

207
IPSS4
8.10 AP: / unit-x / portgroup-y / port-z (ISDN-BA)
8.10.1 AP: / unit-x / portgroup-y / port-z, Overview

2510”.
8.10.2 AP: / unit-x / portgroup-y / port-z, Main (ISDN-BA)

2510”.
8.10.3 AP: / unit-x / portgroup-y / port-z: Configuration (ISDN-BA)
Please note:
An ISDN-BA user port is not configurable if it is member of a hunting group.

207
IPSS4
8.10.3.1 AP: / unit-x / portgroup-y / port-z: Configuration - Port ISDN-BA
Table 82: AP: / unit-x / portgroup-y / port-z: Configuration - Port ISDN-BA

ISDN-BA Port Enable  Enable the ISDN-BA user port.

ISDN-BA Port, Sub- Add… Add a single subscriber identification (MSN num-
scriber Identifica- ber) to the subscriber identifications list. Up to 31
tions ISDN-BA subscriber identifications can be con-
figured.
Subscriber Number 0 … 20 characters Subscriber number of the ISDN-BA user port.

207
IPSS4
Table 82: AP: / unit-x / portgroup-y / port-z: Configuration - Port ISDN-BA (continued)
vice.
The Privacy header is only inserted if “Privacy” is
Id
ISDN-BA Port Register As Global  Expand the contact address field with the coun-

code.
Register Default  Register the default number only or register all
Number only  numbers.
Layer 1 Perma-  With the permanently activated ISDN-BA layer 1,

nected to it.
activated per call.
profiles> list.

207
IPSS4
Table 82: AP: / unit-x / portgroup-y / port-z: Configuration - Port ISDN-BA (continued)
BA profiles> list.
Available Services.
8.10.3.2 AP: / unit-x / portgroup-y / port-z, Configuration - CTP
Table 83: AP: / unit-x / portgroup-y / port-z, Configuration - CTP

The layer rate property of an ISDN-BA user port
is fixed to P0_nc.
The number of timeslots of an ISDN-BA user
port is fixed to 2.
of P0_nc, e.g. 1 … 31.
The timeslot(s) property of an ISDN-BA user port
is empty.
Connected To Connection Index 0 … 65‘535 Connection index.
CTPs
a-End Working

207
IPSS4
Table 83: AP: / unit-x / portgroup-y / port-z, Configuration - CTP (continued)

tion termination point) where the ISDN-BA user
port is connected to, e.g. /unit-1/port-1.
a-End Working
Unidirectional
End.
sion  CAS AIS supervision is not supported for the
ISDN-BA port.
ging  events.
8.10.4 AP: / unit-x / portgroup-y / port-z: Status (ISDN-BA)
8.10.4.1 AP: / unit-x / portgroup-y / port-z, Status - Port Status
Table 84: AP: / unit-x / portgroup-y / port-z, Status - Port Status

User Port Status Admin State Locked User port locking state / administrative state
(X.731), controlled by the linecard.
Shutting Down
Unlocked
Oper State Enabled User port locking state / operational state
(X.731).
Disabled
Please note:

207
IPSS4
8.10.4.2 AP: / unit-x / portgroup-y / port-z, Status - Supplementary Services
Table 85: AP: / unit-x / portgroup-y / port-z, Status - Supplementary Services

Hotline, Subscriber Subscriber Number 0 … 20 characters ISDN-BA subscriber number of the selected port.
List Configured hotline number.
Hotline Number 0 … 20 characters
- No hotline number is configured or the supple-
mentary service is not activated.
Activated  Hotline supplementary service of the ISDN-BA
 subscriber number.
ISDN-GFP Con- Subscriber Number 0 … 20 characters ISDN-BA subscriber number of the selected port.
trolled Network Ser- Configured Call Forwarding Unconditional num-
CFU DivertedTo 0 … 20 characters
vices, Subscriber ber.
Number
List
- No Call Forwarding Unconditional number is
configured or the supplementary service is not
activated.
CFB DivertedTo 0 … 20 characters Configured Call Forwarding Busy number.
Number No Call Forwarding Busy number is configured
-
or the supplementary service is not activated.
CFNR DivertedTo 0 … 20 characters Configured Call Forwarding No Reply number.
Number No Call Forwarding No Reply number is config-
-
ured or the supplementary service is not acti-
vated.
Please note:
The status of ISDN-GFP controlled supplementary services is only available
if GFP was used to activate the Call Forwarding services.
8.10.4.3 AP: / unit-x / portgroup-y / port-z, Status - Registration

207
IPSS4
Table 86: AP: / unit-x / portgroup-y / port-z, Status - Registration

Registration, Sub- Subscriber Number 0 … 20 characters ISDN-BA subscriber number.
scriber List Transient state after (re-)configuration of the
Registration Status ---
port.
Registered The subscriber number is registered.
tering process.
response.
8.10.4.4 AP: / unit-x / portgroup-y / port-z, Status - Ongoing Calls
Table 87: AP: / unit-x / portgroup-y / port-z, Status - Ongoing Calls

Ongoing Calls ISDN-BA, Local 0 … 30 characters Calling number of an ISDN-BA call.
Number
ISDN-BA, Remote 0 … 30 characters Called number of an ISDN-BA call.
Number
ISDN-BA, SIP Call 0 … 40 characters Value of the internal state machine.
Control
ISDN-BA, ISDN-BA 0 … 40 characters Value of the internal state machine.
Call Control
ISDN-BA, Codec 0 … 40 characters Used codec for the call.

207
IPSS4
8.10.4.5 AP: / unit-x / portgroup-y / port-z: Status - CTP
Table 88: AP: / unit-x / portgroup-y / port-z: Status - CTP

Remote CTP tion termination point) where the P0-nc signal is
SF
tional connection.
SF
tional connection.
been selected.
has been selected.
working remote CTP.

207
IPSS4
8.11 AP: / unit-x / portgroup-y / icChannel
8.11.1 AP: / unit-x / portgroup-y / icChannel, Overview

2510”.
8.11.2 AP: / unit-x / portgroup-y / icChannel, Main

2510”.
8.11.3 AP: / unit-x / portgroup-y / icChannel, Configuration
8.11.3.1 AP: / unit-x / portgroup-y / icChannel, Configuration - CTP

207
IPSS4
Table 89: AP: / unit-x / portgroup-y / icChannel, Configuration - CTP

The layer rate property of an ISDN-BA IC-chan-
nel is fixed to P0_nc.
The number of timeslots of an ISDN-BA IC-chan-
nel is fixed to 2.
of P0_nc, e.g. 1 … 31.
The timeslot(s) property of a ISDN-BA IC-chan-
nel is empty.
Connected To Connection Index 0 … 65‘535 Connection index.
CTPs
a-End Working
tion termination point) where the ISDN-BA IC-
channel is connected to, e.g. /unit-3/icChannel.
a-End Working
Unidirectional
End.
sion  CAS AIS supervision is not supported for the IC
channel.
ging  events.
8.11.4 AP: / unit-x / portgroup-y / icChannel, Fault Management
For the a description of the general aspects of the

− “Fault Management - Status”, and
− “Fault Management - Configuration”
2510”. The following table lists the fault causes of the current AP.

207
IPSS4
Table 90: AP: / unit-x / portgroup-y / icChannel, Fault Management

Affecting Severity
ICC Internal Communica- Communica-  Major This alarm is set as a result of a commu-
tion Channel Fault tion Alarm nication failure in the related IC-channel
(i.e. IC-channel communication between
IPSS4 and the related SUIx1 unit).
This alarm is cleared as soon as the IC-
channel communication between the
IPSS4 and the related SUIx1 unit is
available.
→ Action: Check the configuration of the
SUIx1 IC-channel, especially for the cor-
rect bandwidth configuration (128 kbit/s),
the number of ports in the portgroup or
any missing cross connections.
8.11.5 AP: / unit-x / portgroup-y / icChannel, Status
8.11.5.1 AP: / unit-x / portgroup-y / icChannel, Status - CTP
Table 91: AP: / unit-x / portgroup-y / icChannel, Status - CTP

connected to, e.g. /unit-3/icChannel.
SF
tional connection.
connected to, e.g. /unit-4/icChannel.

207
IPSS4
Table 91: AP: / unit-x / portgroup-y / icChannel, Status - CTP (continued)

SF
tional connection.
been selected.
has been selected.
working remote CTP.

207
Annex User Manual
IPSS4
9 Annex
9.1 Associated MileGate documents
Any version(s) and/or release(s) indicated with the below listed document
titles identify the specific state of the software and/or feature set at the crea-
tion time of the present document. If the present document is published as
part of a document collection, the hyperlinks might open a document valid for
a newer version/release. That updated version is valid in the context of all
units and features described in the document collection.
[012] Release Note “MileGate R4C”
[201] System Description “MileGate R4C”
[202] Safety Instructions “Precautions and safety”
[301] User Guide “MileGate 2510 Installation”
[310] User Guide “MileGate 2310 Installation”
[302] User Guide “MileGate 2310/2510”
[323] User Guide “Management Communication”
[354] Quick Guide “ECST”
[355] User Manual “ECST”
[314] User Guide “TDM Services and Cross Connections”
[340] Quick Guide “TDM Services over PDH/SDH”
[341] Quick Guide “Ethernet Switching”
[447] User Manual “COGE5, COGE5-F ci5ne_r1, ci5un_r1”
[808] Technical Bulletin “MileGate CPS Handling Procedures”
[913] Technical Bulletin “SIP / IPSS4 questionnaire MileGate”
[914] Technical Bulletin “SIP provisioning checklist MileGate”
[915] Technical Bulletin “Feature Licences for MileGate”

207
Annex User Manual
IPSS4
9.2 Technical support
Please refer to the KEYMILE Extranet (via http://www.keymile.com) for sup-

port contact information.
9.3 Product training
Training courses are available for a wide range of KEYMILE products and
applications.
For contact information, course descriptions, locations and dates, refer to the
Website: http://www.keymile.com, then select “Services - Training” from the
menu.

207

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User Manual

SIP VoIP Subscriber Media Gateway

Document PEC EN/LZTBU 371 149 RA

Document release MileGate R4C | 23. July 2014

3 Functions and specifications 16

© KEYMILE July 2014 page 3 of 207 EN/LZTBU 371 149 RA

5.6 Direct dialling in and trunk hunting 86

8 User interface reference 127

© KEYMILE July 2014 page 4 of 207 EN/LZTBU 371 149 RA

9.2 Technical support 207

© KEYMILE July 2014 page 5 of 207 EN/LZTBU 371 149 RA

Figure 1: Application example with IPSS4 9

© KEYMILE July 2014 page 6 of 207 EN/LZTBU 371 149 RA

1.1 Precautions and safety

1.2 Symbols and notations

This User Manual uses the following symbols:

Risk of operating trouble!

1.3 Interfaces and circuit categories

Table 1: Electrical interfaces and circuit categories

© KEYMILE July 2014 page 7 of 207 EN/LZTBU 371 149 RA

1.4 Document history

Table 2: Document history

© KEYMILE July 2014 page 8 of 207 EN/LZTBU 371 149 RA

Figure 1: Application example with IPSS4

© KEYMILE July 2014 page 9 of 207 EN/LZTBU 371 149 RA

© KEYMILE July 2014 page 10 of 207 EN/LZTBU 371 149 RA

2.2 Unit view

Figure 2: IPSX4 unit view

© KEYMILE July 2014 page 11 of 207 EN/LZTBU 371 149 RA

2.3 Block diagram

2.3.1 Implementation for PSTN access

Signalling TDM / packet connection

Power RTP stream SIP

GbE star interface

Figure 3: IPSS4 functional block diagram for PSTN access

© KEYMILE July 2014 page 12 of 207 EN/LZTBU 371 149 RA

© KEYMILE July 2014 page 13 of 207 EN/LZTBU 371 149 RA

2.3.2 Implementation for ISDN-BA access

GbE star interface

Figure 4: IPSS4 functional block diagram for ISDN-BA access

© KEYMILE July 2014 page 14 of 207 EN/LZTBU 371 149 RA

© KEYMILE July 2014 page 15 of 207 EN/LZTBU 371 149 RA

3 Functions and specifications

3.1 Feature licences

Table 3: Feature licences relevant for this unit

3.2 Summary of standards

© KEYMILE July 2014 page 16 of 207 EN/LZTBU 371 149 RA

Table 4: Standards (continued)

© KEYMILE July 2014 page 17 of 207 EN/LZTBU 371 149 RA

Table 4: Standards (continued)

© KEYMILE July 2014 page 18 of 207 EN/LZTBU 371 149 RA

3.3 Main functions and specifications

Table 5: Main functions and specifications

© KEYMILE July 2014 page 19 of 207 EN/LZTBU 371 149 RA

Table 5: Main functions and specifications (continued)

© KEYMILE July 2014 page 20 of 207 EN/LZTBU 371 149 RA

Table 5: Main functions and specifications (continued)

3.4 Call progress tones

© KEYMILE July 2014 page 21 of 207 EN/LZTBU 371 149 RA