Sip Isdn Call Flow

SIP ISDN Call Flow SIP_ISDN_Call_Flow
SIP Client VOIP Network Company

Network
Alice Proxy 1 GW 1 PBX C
This call flow diagram was generated with EventStudio Sequence Diagram Designer
(http://www.EventHelix.com/EventStudio).
This article is based on the call flow presented in

http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-pstn-call-flows-02.txt and is reproduced here as per the
copyright statement at the end of this document.
Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk
group.
allocate Alice's PC allocates a port for receiving RTP data. This port
1:Port 49172 number will be included in the SIP Invite.
2:SIP INVITE Alice dials Carol's telephone number (918-555-3333) which is

Calling #, globalized and put into a SIP URI. The message contains
Called #, information about the RTP port number and the supported voice
Contact, codecs.
Media Information
3:SIP 100 Trying Proxy 1 indicates to the SIP client that it is trying to establish
the call.
4:Identify the Gateway that servers Proxy 1 looks up the telephone number and locates the
Carol gateway that serves Carol. Carol is identified by her extension
(444-3333) in the Request-URI sent to GW 1.
5:SIP INVITE The host portion of the Request-URI in the INVITE is used to
identify the context (customer, trunk group, or line) in which the
private number 444-3333 is valid. Otherwise, this INVITE
message could get forwarded by GW 1 and the context of the
digits could become lost and the call unroutable.
6:SIP 100 Trying GW 1 indicates to the Proxy that it is trying to establish the call.
7:Q.931 SETUP The GW routes the call. Since Carol is served by an ISDN PBX,
the Gateway initiates a Q.931 call setup with the PBX.
8:Q.931 CALL PROCEEDING The ISDN PBX responds with Call Proceeding. This message
indicates that the call is in the process of being setup.
9:Q.931 PROGRESS The ISDN PBX passes call progress information to the
Gateway. This message indicates that the called subscriber is
being rung.
10:SIP 180 Ringing The Gateway sends the Ringing indication back to the proxy.
11:SIP 180 Ringing The proxy forwards the ringing indication to Alice's PC.
12:Q.931 CONNECT Carol has answered the call. This results in Q.931 CONNECT
message being sent to the Gateway.
13:Q.931 CONNECT ACK The Gateway replies with Connect Ack.
allocate The Gateway allocates a port for receiving RTP data from
14:Port 3456 Alice's PC. The port information will be passed to originating
subscriber via the "SIP 200 OK" response.
15:SIP 200 OK The Gateway indicates to the Proxy that the call is successful.
Media information The RTP audio receive port information is also passed in this
message.
16:SIP 200 OK The Proxy forwards the message to Alice's PC.
31-May-21 Generated with EventStudio 1

SIP ISDN Call Flow SIP_ISDN_Call_Flow

Network
17:SIP ACK Alice's PC acknowledges the message.
18:SIP ACK The Proxy forwards the ack to the Gateway.
Two way voice is active at this time. Alice and Carol are talking.
Alice Hangs Up with Carol.

19:SIP BYE SIP BYE signals the release of the call.
20:SIP BYE The Bye is forwarded to the Gateway.
21:Q.931 DISCONNECT The Gateway initiates the call release on SS7 side.
22:SIP 200 OK The Gateway acknowledges the BYE to the Proxy with an 200
OK response code.
23:SIP 200 OK The Proxy forwards the ack to Alice's PC.
free
24:Port 49172
25:Q.931 RELEASE The ISDN PBX indicates to the Gateway that it is releasing the
call.
free
26:Port 3456
27:Q.931 RELEASE COMPLETEThe Gateway acknowledges the call release of the call with the
Release Complete message.

SIP ISDN Call Flow with Message Details SIP_ISDN_Call_Flow

Network
This call flow diagram was generated with EventStudio Sequence Diagram Designer
(http://www.EventHelix.com/EventStudio).
Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk
group.
allocate Alice's PC allocates a port for receiving RTP data. This port
1:Port 49172 number will be included in the SIP Invite.
2:SIP INVITE Alice dials Carol's telephone number (918-555-3333) which is

Calling #, globalized and put into a SIP URI. The message contains
Called #, information about the RTP port number and the supported voice
Contact, codecs.
Media Information
INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0

Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:alice@client.a.example.com>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
opaque="", uri="sips:+19185553333@ss1.a.example.com;user=phone",
response="6c792f5c9fa360358b93c7fb826bf550"
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
3:SIP 100 Trying Proxy 1 indicates to the SIP client that it is trying to establish
the call.
SIP/2.0 100 Trying

Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>;tag=9fxced76sl
CSeq: 2 INVITE
Content-Length: 0
4:Identify the Gateway that servers Proxy 1 looks up the telephone number and locates the
Carol gateway that serves Carol. Carol is identified by her extension
(444-3333) in the Request-URI sent to GW 1.
5:SIP INVITE The host portion of the Request-URI in the INVITE is used to
identify the context (customer, trunk group, or line) in which the
private number 444-3333 is valid. Otherwise, this INVITE
message could get forwarded by GW 1 and the context of the
digits could become lost and the call unroutable.
INVITE sips:4443333@gw1.a.example.com SIP/2.0

Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr>
CSeq: 2 INVITE
Contact: <sips:alice@client.a.example.com>
Content-Length: 154


Network
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
6:SIP 100 Trying GW 1 indicates to the Proxy that it is trying to establish the call.
SIP/2.0 100 Trying

Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1;received=192.0.2.111
CSeq: 2 INVITE
Content-Length: 0
7:Q.931 SETUP The GW routes the call. Since Carol is served by an ISDN PBX,
the Gateway initiates a Q.931 call setup with the PBX.
Protocol discriminator=Q.931
Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband)
Called party number:
Type of number unknown
Digits=444-3333
8:Q.931 CALL PROCEEDING The ISDN PBX responds with Call Proceeding. This message
indicates that the call is in the process of being setup.
Message type=CALL PROC
Channel identification=Exclusive B-channel
9:Q.931 PROGRESS The ISDN PBX passes call progress information to the
Gateway. This message indicates that the called subscriber is
being rung.
Message type=PROG
Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband)
10:SIP 180 Ringing The Gateway sends the Ringing indication back to the proxy.
SIP/2.0 180 Ringing

To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Length: 0
11:SIP 180 Ringing The proxy forwards the ringing indication to Alice's PC.
SIP/2.0 180 Ringing

CSeq: 2 INVITE


Network
Content-Length: 0
12:Q.931 CONNECT Carol has answered the call. This results in Q.931 CONNECT
message being sent to the Gateway.
Message type=CONN
13:Q.931 CONNECT ACK The Gateway replies with Connect Ack.
Message type=CONN ACK
allocate The Gateway allocates a port for receiving RTP data from
14:Port 3456 Alice's PC. The port information will be passed to originating
subscriber via the "SIP 200 OK" response.
15:SIP 200 OK The Gateway indicates to the Proxy that the call is successful.
Media information The RTP audio receive port information is also passed in this
message.
SIP/2.0 200 OK
CSeq: 2 INVITE
Content-Length: 144
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
16:SIP 200 OK The Proxy forwards the message to Alice's PC.
SIP/2.0 200 OK
CSeq: 2 INVITE
Content-Length: 144
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
17:SIP ACK Alice's PC acknowledges the message.
ACK sips:4443333@gw1.a.example.com SIP/2.0

Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>


Network
CSeq: 2 ACK
Content-Length: 0
18:SIP ACK The Proxy forwards the ack to the Gateway.
ACK sips:4443333@gw1.a.example.com SIP/2.0

Max-Forwards: 69
CSeq: 2 ACK
Content-Length: 0
Two way voice is active at this time. Alice and Carol are talking.
Alice Hangs Up with Carol.

19:SIP BYE SIP BYE signals the release of the call.
BYE sips:4443333@gw1.a.example.com SIP/2.0

Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
CSeq: 3 BYE
Content-Length: 0
20:SIP BYE The Bye is forwarded to the Gateway.
BYE sips:4443333@gw1.a.example.com SIP/2.0

Max-Forwards: 69
CSeq: 3 BYE
Content-Length: 0
21:Q.931 DISCONNECT The Gateway initiates the call release on SS7 side.
Message type=DISC
Cause=16 (Normal clearing)
22:SIP 200 OK The Gateway acknowledges the BYE to the Proxy with an 200
OK response code.
SIP/2.0 200 OK
CSeq: 3 BYE
Content-Length: 0
23:SIP 200 OK The Proxy forwards the ack to Alice's PC.
SIP/2.0 200 OK

Network
CSeq: 3 BYE
Content-Length: 0
free
24:Port 49172
25:Q.931 RELEASE The ISDN PBX indicates to the Gateway that it is releasing the
call.
Message type=REL
free
26:Port 3456
27:Q.931 RELEASE COMPLETEThe Gateway acknowledges the call release of the call with the
Release Complete message.
Message type=REL COM
Full Copyright Statement
Copyright (c) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to others, and derivative works that comment on or
otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part,
without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies
and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright
notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing
Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or
as required to translate it into languages other than English.

Sip Isdn Call Flow

Uploaded by

Copyright:

Available Formats

Sip Isdn Call Flow

Uploaded by

Document Information

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Sip Isdn Call Flow

Uploaded by

Copyright:

Available Formats

SIP ISDN Call Flow SIP_ISDN_Call_Flow

SIP Client VOIP Network Company

This article is based on the call flow presented in

2:SIP INVITE Alice dials Carol's telephone number (918-555-3333) which is

13:Q.931 CONNECT ACK The Gateway replies with Connect Ack.

16:SIP 200 OK The Proxy forwards the message to Alice's PC.

31-May-21 Generated with EventStudio 1

SIP Client VOIP Network Company

17:SIP ACK Alice's PC acknowledges the message.

18:SIP ACK The Proxy forwards the ack to the Gateway.

Alice Hangs Up with Carol.

20:SIP BYE The Bye is forwarded to the Gateway.

23:SIP 200 OK The Proxy forwards the ack to Alice's PC.

31-May-21 Generated with EventStudio 2

SIP Client VOIP Network Company

2:SIP INVITE Alice dials Carol's telephone number (918-555-3333) which is

INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0

SIP/2.0 100 Trying

INVITE sips:4443333@gw1.a.example.com SIP/2.0

31-May-21 Generated with EventStudio 3

SIP Client VOIP Network Company

SIP/2.0 100 Trying

SIP/2.0 180 Ringing

SIP/2.0 180 Ringing

31-May-21 Generated with EventStudio 4

SIP Client VOIP Network Company

13:Q.931 CONNECT ACK The Gateway replies with Connect Ack.

16:SIP 200 OK The Proxy forwards the message to Alice's PC.

17:SIP ACK Alice's PC acknowledges the message.

ACK sips:4443333@gw1.a.example.com SIP/2.0

31-May-21 Generated with EventStudio 5

SIP Client VOIP Network Company

18:SIP ACK The Proxy forwards the ack to the Gateway.

ACK sips:4443333@gw1.a.example.com SIP/2.0

Alice Hangs Up with Carol.

BYE sips:4443333@gw1.a.example.com SIP/2.0

20:SIP BYE The Bye is forwarded to the Gateway.

BYE sips:4443333@gw1.a.example.com SIP/2.0

23:SIP 200 OK The Proxy forwards the ack to Alice's PC.

SIP Client VOIP Network Company

Full Copyright Statement

Copyright (c) The Internet Society (2003). All Rights Reserved.

31-May-21 Generated with EventStudio 7

You might also like