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 :93971004

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.1912.5 Summary

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

Protocol Stack SIP G.711,G.729

SIP Architecture

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

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

SIP Operation in User Registration

SIP Operation in Proxy Mode Router

SIP Operation in Redirect Mode

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

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

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/2.0 100 Trying

Complete list of response codes

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

SIP Header field

Message Body v=0 o=UserB 2890844527 2890844527 IN IP4 there.com s=Session SDP c=IN IP4 110.111.112.113 t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap: 0 PCMU/8000

SDP Specification Session description v= (protocol version) o= (owner/creator and session identifier). s= (session name) i=* (session information) u=* (URI of description) e=* (email 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

SIP extensions list

PSTN and SIP interworking

Architecture of IP network communication with PSTN

Voice IP Protocol overview

Interworking between PSTN and IP network ISUP call control signaling

SIP-T

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

IP network interworking with PSTN using SIP-T architecture

Translation of SIP-T

SIP-T scenario 1

SIP-T scenario 2

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.

SIP-T scenario 3

PSTN-SIP v.s PSTN-SIP-PSTN INVITE sip:+1-972-555-2222@proxy.carriew.com/user=phone SIP/2.0 Via : SIP/2/0/UDP gw1/carriwe.com:5060 From : sip:+1-314-555-2222@gw1.carrier.woo,.com;user=phone To: sip:+1-972-555-2222@proxy.carriew.com/user=phone Call-ID:12345602@gw1.carrier.com CSeq: 1 INVITE Contact : sip:+1-314-555-2222@gw1.carrier.woo,.com;user=phone Content-Type: application/sdp Content-length: 156 v=0 o= GATEway1 2890844527 2890844527 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

SIP-I

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.1912.5(Interworking between Session Initiation Protocol (SIP) and the Bearer Independent Call Control Protocol or ISDN User Part) Q.1912.5則定義 SIP與BICC/ISUP在介接單元的信號介接程序。

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 24.229 V5.1.0 Profile B complements Profile A, and both of them are intended to support traffic that terminates within the SIP network.

Scenario

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

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

Summary

Protocol summary

SIP interworking capability “profiles”

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.1912.5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。

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.1912.5, 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