1. Presented by Zhi-Hong Guo Instructed by Assistant Professor Quincy Wu
RFC Session Initiation Protocol for Telephones (SIP-T): Context and Architectures
Presented by Zhi-Hong Guo
Instructed by Assistant Professor Quincy Wu

2. Introduction
It is vital for a SIP telephony network to interwork with the PSTN.
An important characteristic of any SIP telephony network is feature transparency with respect to the PSTN.
Another important characteristic of a SIP telephony network is routability of SIP requests.

3. Introduction (cont.)
The SIP-T effort provides a framework for the integration of legacy telephony signaling into SIP messages.
SIP-T provides the above two characteristics through techniques known as 'encapsulation' and 'translation' respectively.

4. PSTN architecture Three components: Customer premises equipment (CPE)
Telephone set, private branch exchanges (PBX)
The transmission facilities
Trunks and subscriber lines
The switching system
Central offices (CO), tandems

5. PSTN architecture (cont.)

6. Call setup and release
A call requires a communications circuit between two subscribers.
The setup and release of connection is triggered by signals.

7. Signaling System No. 7 (SS7)
SS7 is a global standard for telecommunications
defined by ITU-T.
SS7 follows ISO 7 layer architecture.

8. SS7 Protocol Stack
ISUP:For call control,it defines the protocol and procedures used to set-up, manage and release trunk circuits.
Ex: Call setup or release

9. Basic call setup
IAM: Initial Address Message ACM: Address Complete ANM: Answer Message

10. Basic call release
REL: Release Message RLC: Release Complete Message

11. Encapsulation and translation
Encapsulation: Some of SS7 ISUP messages are encapsulated into the SIP message body in order that information necessary for services is not discarded in the SIP request.
Translation: Some critical SS7 ISUP messages are translated into the corresponding SIP headers in order to determine how the SIP request will be routed.

12. SIP-T flows SIP Bridging (PSTN - IP - PSTN)
PSTN origination - IP termination :The terminator has no use for the encapsulated ISUP and ignores it.
IP origination - PSTN termination: The terminating gateway only performs translation.
IP origination - IP termination: This is a case for pure SIP.

13. SIP Bridging (PSTN - IP - PSTN)
VoIP Network
SIP Proxy
PSTN
MGC
SS7 ISUP
SIP
SIP Proxy
PSTN Phone
PSTN
MGC
SS7 ISUP
PSTN Phone

14. Call-flow PSTN Phone PSTN Phone MGC#1 MGC#2 IAM INVITE IAM 100 TRYING
SIP Proxy
IAM
INVITE
IAM
100 TRYING
ACM
18X
ACM
ANM
200 OK
ANM
ACK
CONVERSATION
REL
RLC
BYE
REL
200 OK
RLC

15. PSTN origination - IP termination
VoIP Network
SIP Proxy
SIP Proxy
SIP
SIP
SIP Proxy
SIP Phone
PSTN
MGC
SS7 ISUP
PSTN Phone

16. Call-flow PSTN Phone SIP Phone MGC IAM INVITE INVITE 100 TRYING 18X
SIP Proxy
IAM
INVITE
INVITE
100 TRYING
18X
18X
ACM
200 OK
200 OK
ANM
ACK
ACK
CONVERSATION
REL
BYE
RLC
BYE
200 OK
200 OK

17. IP origination - PSTN termination
VoIP Network
SIP Proxy
PSTN
MGC
SS7 ISUP
SIP
SIP Proxy
SIP Proxy
PSTN Phone
SIP
SIP Phone

18. Call-flow SIP Phone SIP Proxy MGC PSTN Phone INVITE INVITE 100 TRYING
IAM
100 TRYING
ACM
18X
18X
ANM
200 OK
200 OK
ACK
ACK
CONVERSATION
BYE
BYE
REL
200 OK
200 OK
RLC

19. Components of the SIP-T Protocol
Core SIP: defined by RFC 3261
Encapsulation: SIP-T uses multipart MIME bodies to enable SIP messages to contain multiple payloads (SDP,ISUP, etc.).
Translation:
ISUP SIP message mapping
ISUP parameter-SIP header mapping

20. ISUP-SIP message mapping
IAM<->INVITE
ACM<->18X
REL<->BYE

21. ISUP parameter-SIP header mapping
Called Party Number <-> To、Request-URI
The headers of a SIP request (especially an INVITE) may be transformed by intermediaries, and that consequently, the SIP headers and encapsulated ISUP bodies come to express conflicting values effectively.
The SIP headers should take precedence over the ISUP as the contents of SIP headers may be updated in routing within the IP network.

22. SIP content negotiation
Traditionally in SIP, if the terminating device does not support a multipart payload and/or the ISUP MIME type ,it would then reject the SIP request with a 415 Unsupported Media Type specifying the media types it supports.
The originator would have to re-send the SIP request after stripping out the ISUP payload.

23. SIP content negotiation (cont.)
It is highly desirable to have a mechanism by which the originator could signify which bodies are required and which are optional so that the terminator can silently discard optional bodies that it does not understand.
This is upon the terminator having support for a Content-type of multipart/mixed and access to the Content-Disposition header to express criticality.

24. Support for ISUP is optional
UA2 accepts the INVITE irrespective of whether it can process the ISUP.
UA UA2
INVITE-->
(Content-type: multipart/mixed;
Content-type: application/sdp;
Content-disposition: session; handling=required;
Content-type: application/isup;
Content-disposition: signal; handling=optional;)
<--18x

25. Support for ISUP is preferred
UA2 does not support the ISUP and rejects the INVITE with a 415 Unsupported Media Type. UA1 strips off the ISUP and re-sends the INVITE with SDP only.
UA UA2
INVITE-->
(Content-type: multipart/mixed;
Content-type: application/sdp;
Content-disposition: session; handling=required;
Content-type: application/isup;
Content-disposition: signal; handling=required;)
<--415
(Accept: application/sdp)
ACK-->
INVITE--> (Content-type: application/sdp)
<--18x

26. Support for ISUP is mandatory
UA2 does not support the ISUP and rejects the INVITE with a 415 Unsupported Media type. UA1 then directs its request to UA3.
UA UA2
INVITE-->
(Content-type:multipart/mixed;
Content-type: application/sdp;
Content-disposition: session; handling=required;
Content-type: application/isup;
Content-disposition: signal; handling=required;)
<--415
(Accept: application/sdp)
ACK-->

27. Support for ISUP is mandatory (cont.)
UA UA3
INVITE-->
(Content-type: multipart/mixed;
Content-type: application/sdp;
Content-disposition: session; handling=required;
Content-type: application/isup;
Content-disposition: signal; handling=required;)