1. SIP ,SIP-T and SIP-I Protocol
課程 : 新世代網路
教授 : 連耀南
學生姓名 鄭旭鈞
學號 :

2. Outline SIP PSTN and SIP interworking SIP-T SIP-I Summary
SIP introduction
SIP Architecture
Components of SIP
SIP operation mode
SIP message structure
PSTN and SIP interworking
Interworking architecture
Protocol overview
SIP-T
What’s SIP-T
Architecture
Translation of SIP-T
Encapsulation of ISUP in SIP-T
SIP-I
Specification of SIP-I
TRQ. 2815 Q
Summary

3. SIP Introduction
Open, simple, extensible, and lightweight protocol Design for IP Networks
Text-encoded protocol based on elements from the HTTP and SMTP.
Same protocol used between services and call control entities Text-based encoding
Easier to integrate with telephony and Internet functions Supports multiple call legs (i.e., forking) In order to understanding SIP protocol, we need bearing in mind that
VoIP = Signaling +Media

4. Protocol Stack
SIP
G.711,G.729

5. SIP Architecture

6. Components of SIP User Agents: Server Location Server
User Agent Client (UAC) – Send Request (eg. invite…etc)
User Agent Server (UAS) - Responds to clients’ requests (eg. successful ..etc)
Server
Proxy Server
Receive SIP request from UA or other proxy
Forward or proxies the request to another location
Registrar Server
Receive SIP registration request
update user agent’s information
Redirect Server
Receive request from UA or proxy and returns a redirection response(3xx)
Location Server
It contain user URL,IP address script, feature
Routing information about Proxies,gateways and other Location server in SIP network

7. SIP basic operation modes
Proxy Mode
Proxy Server
Peer to Peer Mode
RTP

8. SIP Operation in User Registration

9. SIP Operation in Proxy Mode
Router

10. SIP Operation in Redirect Mode

11. SIP message structure Request = Request-Line *
( general-header | request-header | entity-header )
CRLF
[message-body]
Request-Line = Method SP Request-URI SP SIP-Version CRLF
Response = Status-Line *
( general-header | response-header | entity-header )
[ message-body ]
Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF

12. Request-Line of request message
Request-Line = Method SP Request-URI SP SIP-Version CRLF
Method :
Register : Register to Registrar
Invite : Invite someone to participate in a session
Cancel : Cancel the invitation
Bye : Finish the call
ACK : Request confirm
Options : Query a server to its capabilities
Request-URI = SIP-URL without parameter or header element
SIP-Version = ”SIP/2.0”
EX :
URI-Uniform Resource Identifier

13. Status Line of Response message
Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
SIP-Version = ”SIP/2.0”
Status-Code :
provisional response
1xx – Informational
final response
2xx - Successful
3xx - Redirection
4xx - Request Failure
5xx - Server Failure
6xx - Global Failures
Reason-Phrase = “Trying”, “Ringing”…
Ex : SIP/ Trying

14. Complete list of response codes

15. Request message Headers
general-header
Apply to both request and response messages
-Ex: Via, From, To, Call-ID, CSeq, Contact, User-Agent …
request-header
The additional information about the request
-Ex: Proxy-Authorization, Max-Forwards, …
entity-header
Define meta-info about the message body
-Ex: Content-Length, Content-Type
The resource identified by the request
-Ex: Allow, Expires
*Refer to RFC 2543 Section 4.1

16. SIP Header field

17. Message Body v=0 o=UserB 2890844527 2890844527 IN IP4 there.com
s=Session SDP
c=IN IP
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap: 0 PCMU/8000

18. SDP Specification Session description v= (protocol version)
o= (owner/creator and session identifier).
s= (session name)
i=* (session information)
u=* (URI of description)
e=* ( address)
p=* (phone number)
c=* (connection information - not required if included in all media)
b=* (bandwidth information)
One or more time descriptions (see below)
z=* (time zone adjustments)
k=* (encryption key)
a=* (zero or more session attribute lines)
Zero or more media descriptions (see below)
Time description
t= (time the session is active)
r=* (zero or more repeat times)
Media description
m= (media name and transport address)
i=* (media title)
c=* (connection information - optional if included at session-level)
a=* (zero or more media attribute lines)
*Refer to RFC 2327 Section 6

19. SIP extensions list

20. PSTN and SIP interworking

21. Architecture of IP network communication with PSTN

22. Voice IP Protocol overview

23. Interworking between PSTN and IP network ISUP call control signaling

24. SIP-T

25. What is SIP-T? SIP-T(SIP for Telephones) is define by IETF
Not an extension to SIP – a set of practices for interfacing SIP to the PSTN, It provides two key characteristics
Encapsulation of ISUP in SIP
Translation of ISUP parameters to SIP headers
Implemented at PSTN gateways, and carried end-to-end
SIP-T Specification families :
RFC3372 : SIP for Telephones (SIP-T): Context and Architectures
RFC3398 : ISUP to SIP Mapping
RFC2976 : The Session Initiation Protocol (SIP) INFO Method
RFC3204 : MIME media types for ISUP and QSIG Objects

26. IP network interworking with PSTN using SIP-T architecture

27. Translation of SIP-T

28. SIP-T scenario 1

29. SIP-T scenario 2

30. Encapsulation of ISUP in SIP-T
Extending SIP-T by encoding SS7 ISUP signaling messages allows MGCs using SIP-T to be compatible with the PSTN.
SIP-T encodes and transmits the native signaling messages from one SCN to another. To do this, SIP-T has been extended with MIME encoding of signaling messages. The PSTN signaling messages are appended to the SIP-T messages (such as INVITE, ACK, BYE) using binary encoding.
The use of MIME encoding with content type: APPLICATION allows PSTN signaling messages to be tunneled between MGCs. The use of content SUBTYPE enables SS7 ISUP messages to be differentiated by the receiving MGC.

31. SIP-T scenario 3

32. PSTN-SIP v.s PSTN-SIP-PSTN
INVITE SIP/2.0
Via : SIP/2/0/UDP gw1/carriwe.com:5060
From :
To:
CSeq: 1 INVITE
Contact :
Content-Type: application/sdp
Content-length: 156
v=0
o= GATEway IN IP4 gatewayone.carrier.com
s=Session SDP
c= IN IP4 gatewayone carrier.com
t= 0 0
m= audio 3456 RTP/AVP 0
a = rtpmap:0 PCMU/8000

33. SIP-I

34. Specification of SIP-I
SIP-I (SIPwithEncapsulatedISUP ) is defined by ITU-T (only draft until now)
ITU-T Series Q Supplement 45: Technical Report TRQ.2815(Requirements for Interworking BICC/ISUP Network with Originating/Destination Networks based on Session Initiation Protocol and Session Description Protocol)
定義了SIP與BICC/ISUP 互通時的技術需求，包括閘道器類型、介接單元(Interworking Unit) 所應支援的協定能力配置集與閘道器的安全模型等。
ITU-T Recommendation Q (Interworking between Session Initiation Protocol (SIP) and the Bearer Independent Call Control Protocol or ISDN User Part)
Q 則定義 SIP與BICC/ISUP在介接單元的信號介接程序。

35. TRQ. 2815
Defines the signaling interworking between the Bearer Independent Call Control (BICC) or ISDN User Part (ISUP) protocols and Session Initiation Protocol (SIP) with its associated Session Description Protocol (SDP) at an Interworking Unit (IWU)
Profile C supports the trunking of traffic via transit SIP networks using MIME encoded encapsulatedISUP
(SIP-I)
Profile A was defined to satisfy the demand represented by 3GPP in TA V5.1.0
Profile B complements Profile A, and both of them are intended to support traffic that terminates within the SIP network.

36. Scenario

37. Q.1912.5 ISUP to SIP/SDP mapping Encapsulation Scenario
Message mapping
Parameter mapping
Scope of parameter
Mapping of ISUP parameters to SIP/SDP
Initial address message (IAM) mapping to SIP
Encapsulation
Scenario
Encapsulation format

38. ISUP- SIP-I Message mapping
ISUP Acro
ISUP Message name
SIP message
ACM
Address complete
180 Ringing 183 Session progress (profile C only)
ANM
Answer
200 OK INVITE
APM
Application transport
INFO or 183 Session progress
BLA
Blocking acknowledgement
ISUP side only
BLO
Blocking
CCR
Continuity check request
CFN
Confusion
INFO or 183 Session progress
CGB
Circuit group blocking
BYE 500 Server Internal Error
CGBA
Circuit group blocking ACK
CGU
Circuit group unblocking
CGUA
Circuit group unblocking ACK
CON
Connect
COT
Continuity
UPDATE
CPG
Call progress
CRG
Charge information
CQM
Circuit group query
CQR
Circuit group query response
DRS
Delayed release (reserved – used in 1988 version)
FAA
Facility accepted
INFO 183 Session progress
FAC
Facility
FAR
Facility request
FOT
Forward transfer
FRJ
Facility reject
NOTE: The Release Complete message is a link specific message and may invoke BYE on some legs.
GRA
Circuit group reset acknowledgement
GRS
Circuit group reset
BYE 500 Server Internal Error
IAM
Initial address
INVITE
IDR
Identification request
INFO 183 Session progress
IRS
Identification response
INF
Information
INR
Information request
LPA
Loop back acknowledgement
LOP
Loop prevention
NRM
Network resource management
OLM
Overload
PAM
Pass-along
PRI
Pre-release information
REL
Release
BYE Message codes]
RES
Resume
RLC
Release complete
BYE (note)
RSC
Reset circuit
BYE 500 Server Internal Error
SAM
Subsequent address
SDM
Subsequent directory number
SGM
Segmentation
Reassembled message ncapsulated.
SUS
Suspend
UBL
Unblocking
UBA
Unblocking acknowledgement
UCIC
Unequipped circuit identification code
UPA
User part available
UPT
User part test
ISUP side only
USR
User-to-user information

39. Summary

40. Protocol summary

41. SIP interworking capability “profiles”

42. Conclusion 軟交換與終端之間的控製協議方面，SIP是趨勢； 軟交換與應用服務器之間，SIP是主流;
SIP-I協議族的內容遠遠比SIP-T的內容要豐富。SIP-I協議族不僅包括了基本呼叫的互通，還包括了CLIP、CLIR等補充業務的互通；除了呼叫信令的互通外，還考慮到了資源預留、媒體訊息的轉換等；既有固網軟交換環境下SIP與BICC/ISUP的互通，也有移動3GPPSIP與BICC/ISUP的互通，等等。
SIP-I協議族具有ITU-T標準固有的清晰準確和詳細具體，可操作性非常強。
並且3GPP已經採用Q 作為IMS與PSTN/PLMN互通的最終標準。所以，軟交換SIP域與PSTN的互通應該遵循ITU-T的SIP-I協議族。實際上已經有許多電信運營商最終選擇了SIP-I而放棄了SIP-T。

43. Reference Keith Mainwaring :IP Telephony Migration Challenges
PCC.I-TEL/doc.0202/03, “Next Generation Networks - Standards Overview (September 2003)”
ITUT Signalling Protocols for NGN Signalling Protocols for NGN
RFC3398-ISUP to SIP Mapping
RFC3372-SIP-T-Context and Architectures
ITU-T Recommendation Q , modified
王培元 :新世代電信網路電信級分封電話信號系統之效能模式及分析
中國IT 實驗室 :软交换协议比较和发展趋势
Henry Sinnreich : Delivering VoIP and Multimedia Service with Session Initiation Protocol