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SIP Overview.

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Presentation on theme: "SIP Overview."— Presentation transcript:

1 SIP Overview

2 Contents What is Sip? SIP Benefits SIP message structure
SIP components SIP mobility SIP

3 What is SIP? Sip is signaling protocol defined by the Internet Engineering Task Force (IETF) for initiating, managing and terminating voice and video sessions across packet networks A textual based client-server protocol Application layer protocol It’s not Transport Protocol It’s not QoS Reservation Protocol 현재 대부분의 상용 인터넷폰 서비스에서 채택하고 있는 H.323 protocol을 대체할 목적으로 IETF에서 개발. Protocol의 이해는 message flow를 보면서 이해하는것이.개념잡는데 도움이 된다. SIP

4 SIP history Work began in 1995 in IETF mmusic WG
03/1999: RFC 2543, 153 pages, 6 methods 11/1999: SIP WG formed 11/2000: draft-ietf-sip-rfc2543bis-02, 171 pages, 6 methods SIP

5 SIP benefits Integration with existing protocols
Integration well with Web(Http) and (Smtp) Simplicity Simple protocol vs H.323 Textual based, so parsing generation are simple Extensibility HTTP and SMTP extensibility Modularity Initiation,termination,change,user location, basic registration. Separating notion of a session to invite a user Scalability Off DNS, BGP Server processing. Simplicity http protocol과 많은 유사성을 보이기때문에 http parser를 수정하여 sip에 이용가능. Extensibility: 인터넷이 계속 확장됨에 따라 여러 IP telephony protocols돌이 광범위하게 보급되것이고, 그에따라 서로 호완성과 확장성이 개발초기단계부터 고려되었다. HTTP, SMTP의 확장성을 그대로 물려받음. Scalability Domain: DNS, registrator서버가 location 제공, 메시지 보낼때마다, DNS query할 필요 없다. BGP: location server가, via field 메시지 자체에 라우팅 정보가 들어 있다. SIP – URI에 다음홉의 주소가 들어있다. transaction이 한번 일어나면, Contact필드를 이용하여 호스트까지 다이렉트로 메세지보낸다. Server processing Proxy server는 세션이 한번열결되고 나면, signaling path에서 drop되기 때문에 서버의 유지보수가 편하다. Ingegration MIME conents. -Invite에 대한 redirect response가 html document가 될수 있다. (소리나 그림으로 사용자의 위치를 알려줄수 있다) Invite 메시지에 Applet을 이용하여 input을 받을 수있다. SIP는 URL로 user를 identify 하기 때문에 Web pagesk 안에 sip id가 emdbed될수 있다.그로인해 Web page 에 URL클릭하는것만으로 call을 initiategkftn 있다. -SIP Invite message 로 보내질수 있다. Sip 어드레스랑 아이덴티컬하기때문에,,, 그리고 SIP proxy server는 같은 textbased protocol이기 때문에 쉽게 SIP message를 SMTP message로 Reformat할 수 있다. 즉. 보이스메일과 이메일을 통합한것으로 인비테이션을 보이스메일로 이메일을 통하여 전달하게 할 수 있는것이다. Voice mail. - SIP

6 SIP Messages SIP Requests:
INVITE – Initiates a call by inviting user to participate in session. ACK - Confirms that the client has received a final response to an INVITE request. BYE - Indicates termination of the call. CANCEL - Cancels a pending request. REGISTER – bind a permanent address to current location OPTIONS – Used to query the capabilities of a server. SIP

7 SIP Response Codes 1yz Informational 2yz Success 3yz Redirection
100 Trying 180 Ringing (processed locally) 181 Call is Being Forwarded 2yz Success 200 ok 3yz Redirection 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 4yz Client error 400 Bad Request 401 Unauthorized 404 Not Found 405 Method not Allowed 407 Proxy Authentication Required 415 Unsupported Media Type 482 Loop Detected 486 Busy Here 5yz Server failure 500 Server Internal Error 6yz Global Failure 600 Busy Everywhere SIP

8 Invite Message Example
Request Method Response Status INVITE SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: BigGuy To: LittleGuy Call-ID: CSeq: 1 INVITE Subject: Happy Christmas Contact: BigGuy Content-Type: application/sdp Content-Length: 147 SIP/ OK Via: SIP/2.0/UDP here.com:5060 From: BigGuy To: LittleGuy Call-ID: CSeq: 1 INVITE Subject: Happy Christmas Contact: LittleGuy Content-Type: application/sdp Content-Length: 134 Message Header Fields v=0 o=UserA IN IP4 here.com s=Session SDP c=IN IP t=0 0 m=audio RTP/AVP 0 a=rtpmap:0 PCMU/8000 v=0 o=UserB IN IP4 there.com s=Session SDP c=IN IP t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Payload “receive RTP G.711-encoded audio at :49172” SIP

9 Address Header Fields From: message originator To: final recipient
Request-URI: current destination; may change along signaling path Contact: appears in INVITE / OPTIONS / ACK / REGISTER requests and in responses. It indicates direct response address to which subsequent transactions are sent. A UA may send subsequent BYE or ACK to Contact: address (unless configured to use an outbound proxy). It includes redirection address in 3xx and 485 responses. It includes additional error information in 4xx, 5xx, and 6xx responses. It may include preference weights. It includes current location in REGISTER requests. Multiple Contact: header fields may be included. SIP

10 Session Description Protocol (SDP)
Convey sufficient information to enable participation in a multimedia session SDP includes description of: Session name and purpose Times the session is active Media to use Information where to send and receive media Contact information SIP

11 Session Description Protocol (SDP)
o=sisalem IN IP s=SIP Example c=IN IP v=0 t= m=audio RTP/AVP 0 a=rtpmap:0 PCMU/8000 M = Rtp/avp 0 rtp payload(pcmu) SIP

12 SIP Distributed Architecture
SIP Components Location Server Redirect Server Registrar Server H.323 SIP/H.323 Gateway Proxy Server Proxy Server PSTN User Agent SIP/PSTN Gateway SIP

13 SIP Components User agent(user application) Proxy Server
UA Client(caller) UA Server(called party) Proxy Server Redirect Server Register SIP

14 User Agents An application that initiates, receives and terminates calls. User Agent Clients (UAC) – An entity that initiates a call. User Agent Server (UAS) – An entity that receives a call. Both UAC and UAS can terminate a call. Both SW and HW available SIP

15 Proxy Server An intermediary program that acts as both a server and a client to make requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. Interprets, rewrites or translates a request message before forwarding it. Stateless proxy server No information about the message is stored Stateful proxy server Forking reliability SIP

16 SIP Operation in Proxy Mode
User on left-hand side is initiating a call to on right-hand side; Callee registered with his server previously Location Server Callee #2 #3 INVITE From: To: sip: Call-ID: #4 INVITE From: To: sip: Call-ID: #1 OK 200 From: To: sip: Call-ID: #5 OK 200 From: To: sip: Call-ID: #6 Proxy/Registrar ACK #7 Media stream #8 SIP

17 Location Server A location server is used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s). Locating service LDAP SQL SIP

18 Redirect Server After location service to look up a user, location information is sent back to the caller in a redirection class response, which concludes the transaction. Unlike a proxy server, the redirect server does not forward SIP request. Unlike a user agent server, the redirect server does not accept or terminate calls. SIP

19 SIP Operation in Redirect Mode
Redirect Server/Location Server Callee #2 #3 INVITE #1 302 moved temporarily Contact: #4 ACK #5 Proxy INVITE #6 OK 200 #7 ACK #8 SIP

20 Registrer Server A server that accepts REGISTER requests.
The register server may support authentication. A registrer server is typically co-located with a proxy or redirect server and may offer location services. SIP

21 SIP Registration SIP

22 Simplified SIP Call Setup and Teardown
INVITE Location/Redirect Server Proxy Server Proxy Server User Agent 302 (Moved Temporarily) ACK INVITE Call Setup INVITE 302 (Moved Temporarily) ACK INVITE 180 (Ringing) 180 (Ringing) 180 (Ringing) 200 (OK) 200 (OK) 200 (OK) ACK ACK ACK Media Path RTP MEDIA PATH Call Teardown BYE BYE BYE 200 (OK) 200 (OK) 200 (OK) SIP

23 SIP and Terminal Mobility
Home Network HP REGISTER #2 Cell 2 Signalling REGISTER #1 FP Cell 1 Visited Network SIP

24 SIP and Terminal Mobility
Home Network INVITE #1 HP #4 INVITE #2 Cell 2 Signalling Data FP Cell 1 INVITE #3 Visited Network SIP

25 SIP and Terminal Mobility
Home Network HP reINVITE #3 #1 #4 REGISTER #3 Cell 2 Signalling Data REGISTER #2 FP Cell 1 Visited Network SIP

26 Personal Mobility Invest: SIP


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