SIP Certification Rel.1
SIP Certification Rel.1
SIP Certification Rel.1
Training
o PC-to-Phone (net2phone.com)
IP Network
o Phone-to-Phone (Paegas)
IP Network
o Phone-to-PC as well
o Supporting Protocols
Gateway Location, QoS, inter-domain AAA
(Authentication, Authorization, Accounting), address
translation, IP, etc.
TCP UDP
rt
IPv4, IPv6
k
Data
Link
SIPPING (Session Deals with standardizing extension to SIP protocol that does not have a bearing on the base
SIP protocol - i.e., all SIP peripheral activities (like support of Message Waiting Indicator
Initiation Proposal feature using SIP, SIP-T, ISUP-SIP mapping, SIP Call flows, AAA requirements in SIP etc
Investigation)
SIMPLE (SIP for Instant Deals with standardizing Presence and Instant Messaging (IM) using SIP (E.g., extensions
to MSRP protocol for Session Mode Messaging unlike the original page-mode messaging
Messaging and Presence offered by SIMPLE), PIDF - Presence Info Data format, XCAP – XML Configuration Access
Leveraging Extensions) Protocol etc
MMUSIC (Multiparty Chartered to specify protocol required for Internet conferencing and multimedia
communications. Specifies protocols such as SDP, RTP/RTCP, RTSP, Interactive
Multimedia Session Communication Establishment (ICE) for NAT discovery etc)
Control)
XCON (Centralized The focus of this working group is to develop a standardized suite of protocols for tightly-
coupled multimedia conferences, where strong security and authorization requirements are
Conferencing) integral to the solution. Standardizes protocols (based on SIP) like CPCP (conferencing
policy control protocol), BFCP (binary floor control protocol) etc
MIDCOM (Middle Box Chartered to address NAT/Fire Traversal issues. Standardizes protocols like MIDCOM for
pin-hole management of NAT; STUN (simple traversal of UDP thru NATs) etc
communication)
ENUM (Electronic Deals with converting E.164 numbers to routable URIs (similar to DNS). In fact, ENUM is a
nothing but a glorified DNS for VoIP. It uses the same building blocks of DNS like the
Numbering) NAPTR (Naming Address Pointer) records for specifying the E.164 to URI conversion
PINT (PSTN Interworking) This WG specifies a protocol to perform the corollary of the SPIRITS (described
above).
For instance this specifies SIP URI scheme changes to implement services such as
click-to-call (based on 3rd Party Call Control mechanism). I.e., this deals with
activating services from the Internet and rendering it over the PSTN
JAIN (Java Advanced Developing abstract APIs for developing service creations across PSTN,
ATM, IP, etc.
Intelligent Network)
PARLAY Group Aims to intimately link IT applications with the capabilities of the
telecommunications world by specifying and promoting application
programming interfaces (APIs) that are secure, easy to use, rich in
functionality, and based on open standards.
Parlay integrates telecom network capabilities with IT applications via a
secure, measured, and billable interface.
PSTN GW
PBX
3rd Party
SIP/SIP-T Applicati
ons
DNS Locatio Applicat
n ion
Server Server
OSA
SIP Network Gatew
ay
EBP SIP/SIP ISC
C CMS/G
MTS C SIP SIP/SIP ISC
User Applicati SIP/SIP-T C
Agents on HSS
SCF
Servers
PacketCable Network
PCF
M IM Subsystem
MT MT
A A GCF
WiFi/ V F
WiMa DSL TTH
x
CDMA/
GSM/UMT
S 14
SIP Certification Training 1.1 © Copyright 2006 Wipro Ltd 14
SIP "Trapezoid"
DNS Server Location
Server
DN
S
Outboun SI Inbound
d Proxy P Proxy
Server Server
SI SI
P P
RT
P
User User
Agent A Agent B
SIP Certification Training 1.1 © Copyright 2006 Wipro Ltd 15
15
SIP User Agent (UA)
o User Agent Client (UAC) :
Logical entity
Creates a new request (initiates a new call)
Uses the client transaction state machinery to send request
Role lasts only for the duration of that transaction
subhodeep@192.219.223.160
#2
Callee
DNS Srv Query ? wipro.com #3
Reply : IP Address of wipro.com SIP Server
subhodeep@wipro.com
200 OK 200 OK
#6
From: sip:Caller@sip.com #5
From: sip:Caller@sip.com
To: sip:subhodeep@wipro.com To: sip:subhodeep@wipro.com
Call-ID: 345678@sip.com
PROXYCall-ID: 345678@sip.com
ACK
#7 sip:subhodeep@wipro.com
#8Media Streams
Behavior
Proxies just receive messages, perform Proxies maintain state during entire
routing logic, send messages out transaction; they remember outgoing requests
as well as incoming requests until transaction
is over
Would result in new execution of SIP routing A forking proxy will be stateful
logic for every retransmission (caching routing
results can help reduce the overload)
Callee@home.com
#2
#3
Callee
aller@sip.com
#1
INVITE Callee@example.com
PROXY
#5
ACK Callee@example.com
Callee@home.com
#6
INVITE Callee@home.com
#7200 OK INVITE
#8
ACK Callee@home.com
address subhodeep@wipro.com
and binds this address to user’s
.223.160
#3 SIP/2.0 200 OK
SIP REGISTRAR
(domain register.wipro.com)
PSTN PSTN
VoIP
Network
INVITE (Call-ID#1)
1 INVITE (Call-ID#2)
1
Calling Called
100 Trying
Party 1
Party
100 Trying
SIP 1
180 Ringing 1
Signaling 180 Ringing 1 Signali
& SDP 200 OK
200 OK
1 ng
Signaling 1
ACK
(UDP or 1
1
ACK
TCP)
Bearer
Media Or
RTP Stream Media
(UDP)
Network Layer
Data Layer
Physical Layer
Network Layer
Data Layer
Physical Layer
Data Layer
Physical Layer
Message Header
Start-Line
From: user <sip:from_user@source>
One or more Header fields To: user <sip:to_user@destination>
An Empty Line indicating the Call-ID: localid@host
end of the header fields CSeq: seq# method
An optional Message Body Content−Length: length of body
o Uses the UTF-8 charset (RFC Content−Type:media type of body
2279) Header: parameter ;par1=value ;par2="value"
o Request and Response Blank Line (CR LF)
messages use the basic format
of RFC 2822 V=0
Message Body
o Message and header field o=origin_user timestamp timestamp IN IP4 host
s=session name
syntax is very much identical to
c=IN IP4 media destination address
HTTP/1.1 t=0 0
m= media type port RTP/AVP payload types
v=0 SDP
o=called 536 2337 IN IP4 h3.clddomain.com
s=session_name_1 INFO
c=IN IP4 192.213.229.147
t=0 0
m=audio 3456 RTP/AVP 0 INFO sip:called@dmn.com SIP/2.0
From: sip:caller@clrdomain.com
To: sip:called@clddomain.com
Contact: <sip:called@clddomain.com> SIP
Call−ID: 31415@clrdomain.com
Requests can CSeq: 1 INFO
Content-Length: 0
have headers
and SDP Requests may not
SDP
have SDP
SIP/2.0 200 OK
From: sip:caller@clrdomain.com SIP/2.0 487 Request Rerminated
To: sip:called@clddomain.com From: sip:caller@clrdomain.com
Call−ID: 31415@clrdomain.com To: sip:called@clddomain.com
SIP Call−ID: 31415@clrdomain.com SIP
CSeq: 1 OPTIONS
Accept: application/sdp CSeq: 1 INVITE
Accept-Encoding: gzip Content-Length: 0
Accept-Language: en
Content-Type: application/sdp
Content-Length: 274 SDP
v=0
o=called 536 2337 IN IP4 h3.clddomain.com
s=session_name_1 SDP Provisional – 180 Ringing
c=IN IP4 192.213.229.147
t=0 0
m=audio 3456 RTP/AVP 0 SIP/2.0 180 Ringing
From: sip:caller@clrdomain.com
To: sip:called@clddomain.com
Contact: <sip:called@clddomain.com> SIP
Call−ID: 31415@clrdomain.com
Response can CSeq: 1 INVITE
Content-Length: 0
have headers
and SDP
Response may not SDP
have SDP
From: sip:caller@clrdomain.com
To: sip:called@clddomain.com
Call−ID: 31415@clrdomain.com
CSeq: 1 OPTIONS
Contact: <sip:alice@atlanta.com>;expires=3600
Contact:<sip:alice@chicago.com>
Route: <sip:alice@atlanta.com>,<sip:carol@chicago.com>,
<sip:bob@biloxi.com>
Application - pkcs7-signature
ghyHhHUujhJhjH77n8HHGTrfvbnj756tbB9HG4VQpfyF467GhIGfHfYT6
4VQpfyF467GhIGfHfYT6jH77n8HHGghyHhHUujhJh756tbB9HGTrfvbnj
n8HHGTrfvhJhjH776tbB9HG4VQbnj7567GhIGfHfYT6ghyHhHUujpfyF4
7GhIGfHfYT64VQbnj756
INVITE (SDPo)
1 Location Lookup
2
Lookup Result 3
INVITE (SDPo)
4
100 Trying 5
Session
180 Ringing Initiation
6
180 Ringing
7
200 OK (SDPT)
8
200 OK (SDPT)
9
ACK
10
Session In
Two way Speech Path
Progress
11
BYE
Session
200 OK Teardown
12
100 Trying (CSeq:1 2
100 Trying (CSeq:1 2
INVITE) INVITE)
200 OK (Cseq:1 INVITE) 3 486 Busy Here (CSeq:1 INVITE)
3
4
ACK (CSeq:2 ACK) 4
ACK (CSeq:1 INVITE)
Second
Transaction
BYE (CSeq:3 BYE)
5
200 OK (CSeq:3 BYE) 6
Second
Transaction
BYE (F-Tag: Xxx, T-Tag: Yyy)
5
Reque Reque
st-URI st-URI
DigitMethod
user interface
The URI including all URI parameters is enclosed in "<" and ">“
If no "<" and ">" are present, all parameters after the URI are
header parameters, not URI parameters
o "q" and "expires“ parameters are only used when the
Contact is present in a REGISTER request or response,
or in a 3xx response
o For Request forwarding, targets are processed from
highest q value to lowest, equal q values may be
processed in parallel
o expires" parameter indicates expiration of the URI
100 Trying 100 Trying 2
2
200 OK (SDPT) 3
200 OK (SDPT) 3
ACK 4
ACK (SDPO)
4
INVIT
1xx
E
INVIT status
CANCE 1xx BYE
1xx
E change
200
L 200
Call Proceeding
failure Callee
>= 2xx
Answer
INVITE
status 300 INVITE
max(T1*2n,
status
T2) status
Failure Success
ACK BYE
32s - 200
- ACK Confirmed
-
event BYE
message 200
sent
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SIP State Transition - Client
Initial
7 INVITE
-
sent
-
INVIT
T1*2n E
INVITE
Calling give up
BYE
1xx
o URI Parameters:
transport: Determines the transport mechanism to be
used for sending SIP messages (i.e. UDP, TCP, TLS,
SCTP)
maddr: Indicates the server address to be contacted for
this user, overriding any address derived from the host
field
ttl: Determines the time-to-live value of the UDP
multicast packet
lr: Indicates that the element responsible for this
resource implements the loose routing mechanisms -
used in the Record-Route header
sip:+919845202688@airtel.kk.com:5060;use
r=phone?Subject=SIP
DNS-Server
Query
1.3.1.9.5.8.6.8.6.4.e164.arpa.?
Response
sip:ssarkar@wipro.com
“Call setup”
Dial SIP
+4686859131 sip:ssarkar@wipro.com
Gateway
SIP Server
Addres Indicates
s Spec Loose
Routing
Record-Route: <sip:server10.biloxi.com;lr>
BYE BYE
BYE
200 OK
200 OK
200 OK
Addres Indicates
s Spec Loose
Routing
Route: <sip:server10.biloxi.com;lr>
A B C D
INVITE B
INVITE C
Route C,D Route D INVITE D
A B C D
INVITE D
INVITE D
Route B,C Route C INVITE D
Via:172.16.16.160
Via:172.16.16.120
Via:172.16.16.120 Via:192.219.223.160
Via:192.219.223.160
Via:192.219.223.160
Request
Response
originating host
v=0
o=alice 2890844526 2890844526 IN IP4
host.anywhere.com
s=
c=IN IP4 host.anywhere.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 51372 RTP/AVP 31
Answered SDP
a=rtpmap:31 H261/90000
m=video 53000 RTP/AVP 32 v=0
a=rtpmap:32 MPV/90000 o=bob 2890844730 2890844730 IN IP4
host.example.com s=
c=IN IP4 host.example.com
t=0 0
m=audio 49920 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 0 RTP/AVP 31
m=video 53000 RTP/AVP 32
a=rtpmap:32 MPV/90000
Backwards Speech Path (audible ringing)
PRACK
PRACK
200 OK
200 OK
UPDATE (6) 8
200 OK (Answer 3)
o User B generates an 200 OK 9
UPDATE request (7) with an ACK
offer
o User A answer is sent in the
200 response (8)
o Finally, User B answers the
call, resulting in a 200 OK
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SIP – Event Notification Framework
o Provide an extensible framework by which SIP nodes can
request notification from remote nodes indicating that certain
events have occurred
o Examples of such services include automatic callback
services (based on terminal state events), buddy lists (based
on user presence events), message waiting indications
(based on mailbox state change events)
o Entities in the network can subscribe to resource or call state
for various resources or calls in the network, and those
entities (or entities acting on their behalf) can send
notifications when those states change
o Defines couple of new METHODs for this purpose:
SUBSCRIBE
NOTIFY
NOTIFY (SubscriptionState: Active) 3
200 OK Zxx Event
4
Occurred
NOTIFY (SubscriptionState: Active) 5
200 OK Zxx Event
6 Occurred
NOTIFY (SubscriptionState: Active)
Terminate
7
Subscription 8
200 OK
9
SUBSCRIBE (Event: Zxx, Expires:0)
200 OK 10
NOTIFY (SubscriptionState: Terminated) 11
12
200 OK
Authentication
Confidentiality
PSK Pre-Shared Keys
Integrity
PKI Public Key Infrastructure
Once the TLS session is set up, the normal call setup
will continue from a.example.com to b.example.com,
with the URI has a SIPS URL and that the Via
indicates that TLS was used
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HTTP Digest Authentication
o Provides a simple challenge-response authentication mechanism (using
a nonce value ) used by a server to challenge a client request (at least
one challenge applicable to the requested resource) and by a client to
provide authentication information
401 (Unauthorized) response message is used by an origin server to
challenge the authorization of a user agent, include a WWW-Authenticate
header field
407 (Proxy Authentication Required) response message is used by a proxy to
challenge the authorization of a client, include a Proxy- Authenticate header
field
o Transmits an MD5 or SHA-1 digest of both the secret password and a
random challenge string (i.e., nonce value) in place of the vulnerable
password in clear text
o Valid response contains a checksum of the username, the password, the
given nonce value, the HTTP method, and the requested URI
o Drawbacks
Authenticating a request to more than one element is problematic
• Leaks hash to elements in the path
Only good for authenticating to the first hop
Authentication Methods:
Authentication
Confidentiality
PSK Pre-Shared Keys
Integrity
PKI Public Key Infrastructure
Public Private
Private
SIP Proxy
RTP/RTCP Media
Firewall/NAT
Firewall/NAT
Media
Signaling
8 5
1
2 3 6 7
10
12
Voice
User Gateway
12
Agent
RTP Relay
o RTP Relay (TURN - Traversal Using Relay NAT) acts as the second
endpoint to each of the actual endpoints that are attempting to
communicate with each other
o A server in the middle of the SIP flow that would manipulate the SDP in
such a way as to instruct the endpoints to send RTP to the Relay instead
of directly to each other
o RTP Relay set up its own internal mapping of a session, noting the
source IP:port of each endpoint sending it RTP packets
o Uses that mapping to forward the RTP from endpoint to endpoint
Diameter
Server
DI
T ER AM
E ET
AM ER
DI
SIP SIP
Clie Clie
nt SI SI nt
P SIP P
SIP
IP Network SIP
DI R Server
Server AM E
ET ET
ER AM
DI
Diameter
Subscriber
Locator
6
MAA
401 Unauthorized
7
401 Unauthorized Successfully
8
REGISTER authenticates the user
UAR
9
10
UAA 11 Successfully
12
REGISTER authenticates the user
MAR 13
14
MAA
200 OK 15
200 OK
16
8
INVITE
200 OK 9
200 OK
16
A to B
when resources in its "send" S
RSVP
setup
RESV S
E E
direction are available, because it R RSVCONF
R
will receive RESV messages from V
A 200 OK V
A 4
the network. However, it does not T PATH T
know the status of the reservations
B to A
I
RSVP
RESV
setup
I
O
in the other direction. B requests N RSVCONF
O
N
confirmation for resource UPDATE (SDP3)
reservations in its "recv" direction 5
A to B
S
o SDP3: When A receives RESV
RSVP
setup
RESV S
E E
messages, it sends an updated R RSVCONF
R
offer (5) to B: V
A 200 OK V
A 4
m=audio 20000 RTP/AVP 0 T PATH T
B to A
I
RSVP
RESV
setup
c=IN IP4 192.0.2.1 O
I
O
a=curr:qos e2e send N RSVCONF
N
a=des:qos mandatory e2e sendrecv UPDATE (SDP3)
5
o SDP4: B responds with an answer
(6) which contains the current 200 OK (SDP4) 6
status of the resource reservation 180 Ringing
(i.e., sendrecv): 7
Bandwidth
Notificati Broker
ons ( Policy Decision
Even Point)
ts Configur
ation
Comman
Edge Router ds
(Policy
Enforcement
Point)
o Trigger events, notifications, and configuration commands
are asynchronous
o More scalable
o Not flexible - difficult to handle modification of
configurations
o Not explicitly customized to handle dynamic DiffServ QoS
Bandwidth
Query (2) Broker
Trigger ( Policy Decision
Events Point)
(1) Response
(3)
Edge Router
(Policy
Enforcement
Point)
o Trigger events generates queries and responses
o Interface between QoS client and provider
QoS client
• Sends QoS reservation requests to the provider
QoS provider
• Accepting or rejecting the request
Only outsourcing
P S
PDP PEP O PDP PEP
C
QoS-Enabled
SIP Client Client SIP
Clie Network Network Network Clie
nt S Access Edge Access Edge CO nt
O P Router Router PS
C
SI SI
P P
Q-SIP
Q-SIP Q-SIP
Server Server
INVITE
INVITE
1
2
3
INVITE
180 Ringing
180 Ringing 4
180 Ringing 6
5
200 OK INVITE 7
COPS REQ 8
9
COPS DEC
200 OK INVITE 10
11
COPS REQ
COPS DEC 12
200 OK INVITE13
ACK
14 ACK
15
16
ACK
INVITE
INVITE
1
2
3
INVITE
180 Ringing
180 Ringing 4
180 Ringing 6
5
200 OK INVITE 7
200 OK INVITE 8
9
COPS REQ
COPS DEC 10
200 OK INVITE11
ACK
12 ACK
13
14
ACK
PSTN GW
PBX
3rd Party
SIP/SIP-T Applicati
ons
DNS Locatio Applicat
n ion
Server Server
OSA
SIP Network Gatew
ay
EBP SIP/SIP ISC
C CMS/G
MTS C SIP SIP/SIP ISC
User Applicati SIP/SIP-T C
Agents on HSS
SCF
Servers
PacketCable Network
PCF
M IM Subsystem
MT MT
A A GCF
WiFi/ V F
WiMa DSL TTH
x
CDMA/
GSM/UMT
S 191
SIP Certification Training 1.1 © Copyright 2006 Wipro Ltd 191
SIP-T Call (SIP Bridging)
180 Ring. (ACM, tSDP) ACM 5
ACM 7
6
Backwards Speech Path (audible ringing)
ANM
200 OK (ANM) 8
ANM 10
9
11
AC
K
Two way Speech Path
REL
BYE (REL) 12
REL 13
14 RLC
200 OK 15
17 RLC 16
180 Ringing
180 Ringing 5
ACM
6
7
Backwards Speech Path
(audible ringing) 200 OK (tSDP)
200 OK (tSDP)
8
9
ACK
ANM
10
12 11 ACK
Two way Speech Path
BYE
BYE 13
REL 15
14
200 OK
17 RLC 16
18 200 OK
Backwards Speech Path (audible ringing)
ANM
200 OK 8
200 OK 10
9
ACK
11
12 ACK
Two way Speech Path
REL
BYE
13
BYE 14
15 RLC
200 OK
17
16
18 200 OK
Content-Type: application/ISUP;
version=nxv3; base=etsi121
Base (etsi121)
Optional Version (X-
NetxProprietaryISUPv3)
MIME Type (application/ISUP)
SIP Header (Content-Type)
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197
Content-Type Header
Content-Type = ( "Content-Type" / "c" ) HCOLON media-type
media-type = "text" / "image" / "audio" / "video" / "application" /
"message" / "multipart" SLASH m-subtype *(SEMI
m-parameter)
o Indicates the media type of the message-body sent to the
recipient
o Must be present if the body is not empty
o If the body is empty, and header field is present, it
indicates that the body of the specific type has zero length
o Compact form of the header field is c
o Examples of valid Content-Type header fields:
Content-Type: application/sdp
c: application/ISUP; version=nxv3; base=etsi121
SIP
Profile
SIP 3GPP
A
Mobile
Network
PSTN/IS
DN PSTN/ISDN
SIP
Profile C
SIP
Profile
SIP
B
Terminating
Network
SIP ISUP
Privacy header field absent PI : Presentation Allowed
ISUP SIP
PI : Presentation Allowed No Privacy header
SIP ISUP
200 OK INVITE Profile A : ANM
Profile B : ANM
Profile C : Generated ANM from the encapsulated ISUP
ISUP SIP
ANM Profile A : 200 OK INVITE
Profile B : 200 OK INVITE
Profile C : 200 OK INVITE with encapsulated ANM
INVITE IAM
INVITE
200 OK (SDPO)
100 Trying
ACK (Hold SDP) REL (16)
INVITE
CANCEL (Q.8650:16)
486 Busy Here
200 OK
ACK
BYE (SIP:486)
200 OK
ACK (SDPo) ACK (SDPo)
Early Media Session
ACK ACK
PRACK (SDPo) PRACK (SDPo)
Early Media Session
INFO (SUS) SUS
SUS
200 OK
INFO (RES) RES
RES
200 OK
Backwards Speech Path (audible ringing)
Two way Speech Path
PSTN GW
PBX
3rd Party
SIP/SIP-T Applicati
ons
DNS Locatio Applicat
n ion
Server Server
OSA
SIP Network Gatew
ay
EBP SIP/SIP ISC
C CMS/G
MTS C SIP SIP/SIP ISC
User Applicati SIP/SIP-T C
Agents on HSS
SCF
Servers
PacketCable Network
PCF
M IM Subsystem
MT MT
A A GCF
WiFi/ V F
WiMa DSL TTH
x
CDMA/
GSM/UMT
S 244
SIP Certification Training 1.1 © Copyright 2006 Wipro Ltd 244
Background
o CableLabs-led initiative that is aimed at developing
interoperable interface specifications for delivering
advanced, real-time multimedia services over two-way cable
plant
o Built on top of cable modem infrastructure, PacketCable
networks
o Use IP technology to enable a wide range of multimedia
services
IP telephony
Multimedia conferencing
Interactive gaming
o Distributed signaling paradigm is SIP (PacketCable 1.1)
o Protocols and architecture developed for DOCSIS-based
cable, but applicable to other broadband access network
technologies
DCS- DCS-
Proxy+GC Proxy+GC
Announcement
Server
P
MT Media Terminal Call State STN
A Adaptor
M Cable Modem Connection State
NCS/TGCP DCS/SIP
Translation, Congestion Control, PSTN,
DB access, Event recording, Routing
Call
Call Agent DCS-Proxy
Signali
ng
QoS
Gate Controller
Signali
ng
DQoS
COPS
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Service Provider Requirements
o Need for differentiated QoS is fundamental
Must support resource reservation and admission control
SIP enables lots of new services; also desire to meet needs of
current users
o Allow for authentication and authorization on a call-by-call
basis
o Need to guarantee privacy and accuracy of feature
information (e.g. Caller ID, Caller ID-block, Calling Name,
Called Party)
o Protect the network from fraud and theft of service
o Must be able to operate in large scale, cost-effectively
End-points keep state associated with their own calls, and not
proxies
3
ACK Verification function)
INVITE (BLV)
4
INVITE (BLV)
5
183 Session Progress
183 Session Progress7 6
PRACK Allocate
8
network
200 OK 9 resources
UPDATE
10
200 OK 11
Commit to
200 OK INVITE network
200 OK INVITE 12 resources
13
ACK
14
Busy Line Verification in Progress (one-way data transfer from MTA to Operator)
Busy Line Verification in Progress (one-way data transfer from MTA to Operator)
P-DCS-OSPS : EI
(Indicates a change to
15
NTFY Emergency Interrupt)
Interrupt
INVITE (EI)
16
200 OK INVITE16
ACK
14
PSTN GW
PBX
3rd Party
SIP/SIP-T Applicati
ons
DNS Locatio Applicat
n ion
Server Server
OSA
SIP Network Gatew
ay
EBP SIP/SIP ISC
C CMS/G
MTS C SIP SIP/SIP ISC
User Applicati SIP/SIP-T C
Agents on HSS
SCF
Servers
PacketCable Network
PCF
M IM Subsystem
MT MT
A A GCF
WiFi/ V F
WiMa DSL TTH
x
CDMA/
GSM/UMT
S 265
SIP Certification Training 1.1 © Copyright 2006 Wipro Ltd 265
IP Multimedia Subsystem (IMS)
Original (late ’90s/early ’00s) definition per 3GPP TS 23.228:
The IP Multimedia CN subsystem comprises all CN elements for provision of
multimedia services. This includes the collection of signaling and bearer related
network elements…
User Ids
User profile
security roaming
Inter-
QoS Working
policy CS/PSTN
SIP
control
Basic
charging Call
Control
Service logic
APIs
CSCF, BGCF
o Session Control Layer
MGCF & MRFC
End Point Registration
Session setup HSS
QoS establishment
Go Gi
Gi
SIP
H.248
IN
Transport and Access
IP
End Point Media SG
H.248
Signalling
Layer Server Converter
SIP
Media
Gatewa
y
SS7
PSTN
Project/3rd Generation
Partnership Project 2)
IETF (Internet Engineering Provide the definitions for SIP, SDP and other protocols underlying IMS
Task Force) IMS is driving some of the work in IETF
OMA (Open Mobile Alliance) Defining services for IMS architecture, e.g. Instant Messaging, Push-to-Talk
ETSI (European TISPAN - TISPAN is merger of TIPHON (VoIP) and SPAN (fixed networks)
Telecommunications Agreement on reuse of 3GPP/3GPP2 IMS in comprehensive NGN plans
Standards Institute)
ANSI (American National Provides protocol definitions used by IMS
Standards Institute) T1.679 covers interworking between ANSI ISUP and SIP
ATIS (Alliance for Addressing end-to-end solutions over wireline and wireless
Telecommunications Industry Nearing agreement to use 3GPP/3GPP2 IMS
Solutions)
Public I
User
Identity
Private I
User
IMS Identity Public II
Subscript User
ion Identity
Private II
User
Identity
Public III
User
Identity
HSS
Location Profile
4 Cx-Query Cx-
3Cx-Query 6 Cx- 7
Resp Pull/Put
8 200 OK Pull/Put Resp
I-CSCF S-CSCF
5 REGISTE
R
2 REGISTE 9 200 OK
R
Visite P-CSCF
d
1 REGISTE 10 200 OK
R GGSN
SGSN
Radio Access Network
4 Cx-Query 3Cx-Query
Resp
5 INVI
S-CSCF TE I-CSCF
14 200 INVITE
6 INVI 13 200 2
OK sip:info@vis
TE OK ited
Visite
INVI
10 IAM d 9
MGCF/ TE I-CSCF
15 200
T-SGW OK
11 ANM 12 200
P OK
7Cx-Query 8 Cx-Query P-CSCF
STN MGW Resp
17 Medi 16 200
a OK
GGSN 1 INVITE
SGSN tel:1411
HSS
Radio Access Network
and finds a match with the INVITE Filter Criteria X to AS1 SPT HeaderA priority 1
filter criteria Y. S-CSCF Header =A Filter criteria Y to AS2 SPT HeaderB priority 2
Terminal convergence
Circuit Switch
Circuit Switch Circuit Switch
Circuit Transport Capability
Transport Capability Access Capability
Switch
Access
Capability
Packet Switch
Transport Packet Switch Packet Switch
Capability Access Capability Transport Capability
xDSL / FTTx
Circuit Switch
Circuit Switch Transport Capability
Access Capability
VoIP / VToA
n n n
Adapter
Radio Access
Network
Base
Station
UMA-enabled Controller GSM/IMS Core
Dual Mode Service
Handset Architecture
WiFi
Tunneled
IMS stack GANC
(UMA)
Network
IP Access Network Controller
RG
VOIP
SIP SIP Server
Fixed/Wireless Telephone
Other Networks
ICF
IP Multimedia
Component (Core IMS)
(SIP based)
PSTN / ISDN Emulation
(SIP-I based)
“Gq”
PSTN / ISDN
Legacy interface
Legacy Terminals
Terminals
Network Attachment Resource and Admission
GW Functionality Control Functionality
NASS RACS
Legacy GW “Go”
Terminals interface TGW
NGN
Terminals Customer MBG
Networks IP Core transport
Access Transport
Network Network
NGN
Terminals
3GPP IP-CAN
3GPP Terminals
RACF
Transport
CPE
Functions
Other NGNs
M-PDF Resource Mediation I-PDF
Rq
Network Rq Rq Rq
Access
A-TRCF C-TRCF I-TRCF
Attachment Access Core
Functions Ub
Go’ Rc Rc
Re Rc Re
Borde
r
Node Border Intermed Acces
Borde Gatew iate s
Inter-TE
NGN Core Network r ay Node Node
Node Node
Access
Network
Home Home
Network Network
Home Home
INVITE Re-
Proxy Proxy
INVITE
#5
#3
E
IST
#6
E
#2
IST
Re-R
#2
REG
ITE
IT E
#4
INVR
REG
#4
IN V
#6
#3
Re-
#5
I N INVITE Cell
VI Cell REGISTE
RE TE 2
2 R
#1
Visited GIS Visited
Proxy #1R TE Proxy
Cell Cell
Visited 1 Visited 1
Network Network
o Watcher er)
Client of the system that asks for information about another user
in the system
o Presentity
User of the system that a watcher can ask about
o Presence Agent (PA)
Purely logical entity
Knows presence state of user
Receives SUBSCRIBE requests
Generates NOTIFY requests
Co-located with proxy/registrar or User Agent
Hello World
1
INVITE
Call 200 OK 2
Transfer
Initiated 3
ACK
using Call
REFER Two way Speech Path
Transfer
4
REFER: ReferTo: C Success
using
202 Accepted 5 REFER
6
INVITE
200 OK
NOTIFY : 200 OK 7
8
ACK
10
200 OK 9
Two way Speech Path
BYE 11
12
200 OK
REFER: ReferTo: <Final
Recipient?Replaces:CallID:2;
FromTag=33 ;ToTag=44
CallID:1;FromTag=11;ToTag=22
202 Accepted
NOTIFY : 200 OK 200 OK CallID:3;FromTag=55;ToTag=66
Call
ID:1;FromTag=11;ToTag=22 ACK CallID:3;FromTag=55;ToTag=66
200 OK
BYE Two way Speech Path
Call
ID:1;FromTag=11;ToTag=22
200 OK
BYE Call
ID:2;FromTag=33;ToTag=44
200 OK CallID:2;FromTag=33;ToTag=44
S
PSTN Swit
MDI VoiceM
User ch
ail
A Server
SIP IP SIP
P Swit VoiceM
User STN ch Networ ail
SIP
A k Server
Switch can
act as a SIP SIP SIP-Based Voice
UA on behalf
Clie Mail System
of TDM
User nt
clients
B
User A calls User B (CFD) forwarded to Voice Mail Server. User A leaves a message for User B and disconnects the call
User B NOTIFY : MWT=YES
receives MWT
“YES” 200 OK
Notification
User B
receives MWT
NOTIFY : MWT=NO
“NO” 200 OK
Notification
Pre-Paid Client
PSTN
Membe
rA Wireless
Network Membe
rE
Membe
rB
Membe
rD
Membe
rC
1
INVITE
2
INVITE Invitations to
invited POC
subscriber
First
ALERTING 3 ALERTING
ALERTING 4
Response
OK First accepted
7
invitation
OK 8
10 Media
4
INVITE
INVITE Invitations to
5
the group
members
First
6
ALERTING ALERTING
Response
ALERTING 7
First
8
ALERTING accepted
ALERTING 11
OK invitation
ALERTING
9
10 OK 12
13
OK
OK
OK
14
15
Talk Burst Confirms
16
4
INVITE
INVITE Invitations to
5
the group
members
First
6
ALERTING ALERTING
Response
ALERTING 7
First
8
ALERTING accepted
ALERTING 11
OK invitation
ALERTING
9
10 OK 12
13
OK
OK
OK
14
15
Talk Burst Confirms
16
createListe getInstanc
SIP Factory ner() SIP Listener e() SIP Factory
createStack(
)
Event
Registrat createProvid
ion er()
SIP Provider
SIP Stack
Proprietary Proprietary
Network
SIP Stack SIP Stack
SipListener SipListener
6
ACK ACK
8
HTTP GET 7
200 OK 9
HTTP POST
10
11
SUBSCRIBE
200 OK
200 OK 12
13
NOTIFY 14
15
200 OK
INVITE 16
17
200 OK
ACK 18
19
INVITE
200 OK
INVITE 21
20
22
200 OK
23
ACK
ACK 23
RTP
SIP Certification Training 1.1 © Copyright 2006 Wipro Ltd 357
357
SIP Lite
o An abstracted view of the SIP protocol that provides a SIP
programming environment for developers
o API specification is primarily developed for the J2SE
platform to provide a rich object model that may be
suitable for midsize devices with more processing power
and memory than mobile handsets, i.e. PDA’s and SIP
phones
o Defines a three-tier architecture, where the Listener exists
for a Dialog, a Call and a CallProvider - listen for incoming
messages, dialogs and calls respectively
o Define a single Message interface identified based on
Request and Response constants
o Defines the concept of a Call and Dialog interface within
which a Call may contain multiple Dialogs
o Specification designed specifically for User Agent
applications
SIP
Lite