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SIP 기술 개요 및 현황

SIP 기술 개요 및 현황. 한국전자통신연구원 표준연구센터 현 욱. SIP Overview. 응용 계층 시그널링 프로토콜 멀티미디어 세션 설정 , 수정 , 종료를 위해 사용 하위 계층 전송 프로토콜과 독립적 UDP, TCP, SCTP Secure transport: TLS over TCP, IPSec HTTP 기반 텍스트 기반 프로토콜 URIs (Uniform Resource Indicators) 사용 SIP-URI 사용  sip:sunok@etri.re.kr

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SIP 기술 개요 및 현황

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  1. SIP 기술 개요 및 현황 한국전자통신연구원 표준연구센터 현 욱

  2. SIP Overview • 응용 계층 시그널링 프로토콜 • 멀티미디어 세션 설정, 수정, 종료를 위해 사용 • 하위 계층 전송 프로토콜과 독립적 • UDP, TCP, SCTP • Secure transport: TLS over TCP, IPSec • HTTP 기반 • 텍스트 기반 프로토콜 • URIs (Uniform Resource Indicators) 사용 • SIP-URI 사용  sip:sunok@etri.re.kr • Personal Mobility 제공 • 동일한 SIP 주소, 다른 위치 (단말) • 현재 사용자의 위치 등록, 수정, 삭제, 검색 기능 • 메시지 포킹(forking) 기능 제공 • 다양한 응용에 활용 가능 • Voice, video, gaming, instant messaging, presence, call control, etc.

  3. SIP Timeline • 1996 • Mark Handley’s SIP(Session Invitation Protocol) • Henning Schulzrinne’s SCIP(Simple Conference Control Protocol) • 1999.3 : IETF MMUSIC WG에 의해 RFC 2543제정 • 1999.9 : IETF SIP WG 설립 • 2000~2002 : RFC 2543bis-01 ~ bis-09 • 2000.6 : RFC 2543bis-01 ••• • 2001.3 : RFC 2543bis-03 ••• • 2002.2 : RFC 2543bis-09 • 2002.7 : RFC 3261표준 제정

  4. SIP Timeline • 2000.12 : SIMPLE WG • SIP-based IMPP • 2001.3 : SIP WG과 SIPPING WG으로 분리 • SIPPING: SIP Proposal Investigation • 2003.7 : XCON WG • Centralized Multimedia Conferencing

  5. SIP related WGs • SDP Extensions • SDPng MMUSIC WG • SIP Core Spec. Maintenance • SIP Protocol Extensions SIP WG • SIP Requirements • Specific SIP Application Services SIPPING WG • SIP for Presence and • Instant Messaging SIMPLE WG

  6. SIP related WGs 1999.9 MMUSIC WG 2001.3 SIP WG SIPPPING WG SIMPLE WG 2000.12

  7. RFCs related to SIP • Base spec • RFC 3261 : SIP : Session Initiation Protocol • RFC 3263 : Locating SIP Servers • RFC 3264 : An Offer/Answer Model with SDP • Extended Features • RFC 2976 : The SIP INFO Method • RFC 3262 : Reliability of Provisional Responses in SIP • RFC 3265 : SIP-Specific Event Notification • RFC 3311 : The Session Initiation Protocol UPDATE Method • RFC 3315 : The Session Initiation Protocol (SIP) Refer Method • RFC 3326 : The Reason Header Field for the Session Initiation Protocol (SIP) • RFC 3327 : Session Initiation Protocol Extension for • Registering Non-Adjacent Contacts • RFC 3428 : Session Initiation Protocol Extension for Instant Messaging

  8. SIP Signaling Flow A B INVITE Create MS, dialog Ringing Prepare MS; Early dialog OK ACK Establish MS, dialog MS in progress Media Streams Terminate MS MS in progress BYE OK Terminate MS; Destroy dialog Destroy dialog

  9. SIP Redirect Model Location Server Request SIP Redirect Server Response INVITE 302 Moved ACK INVITE … SIP Client (UAC:User Agent Client) SIP Client (User Agent Server)

  10. SIP Proxy Model(1/2) Location Server Request SIP Proxy Server Response INVITE INVITE 100 Trying … ACK SIP Client (UAC:User Agent Client) SIP Client (User Agent Server)

  11. SIP Proxy Model (2/2) Location Server • Forking Request SIP Proxy Server Response INVITE INVITE … 100 Trying INVITE ACK … SIP Client (User Agent Server) SIP Client (UAC:User Agent Client) SIP Client (User Agent Server)

  12. SIP Components • UAC (User Agent Client) • SIP 요청 메시지를 생성하는 논리적 구성요소 • SIP transaction을 개시하며, 해당 transaction 존속기간 동안 UAC로 동작 • UAS (User Agent Server) • 수신한 SIP 요청 메시지에 대한 응답 메시지를 생성하는 논리적 구성요소 • 요청 메시지 수용, 거절, Redirect • UA (User Agent) = UAC + UAS • Registrar • REGISTER 메시지를 통해 사용자가 등록시킨 사용자 접속주소 저장 • 특정 사용자로의 접속주소에 대한 정보 제공

  13. SIP Components • Proxy Server • UAC와 UAS 사이에서 SIP 메시지 라우팅을 담당하는 서버 • 메시지 처리를 위해 UAC, UAS로써 동작하며, 경우에 따라 수신 메시지 수정 • Stateful Proxy/Stateless Proxy • Redirect Server • 요청 메시지에 대한 3xx 응답을 생성하는 UAS • 3xx 응답을 통해 클라이언트 접속주소를 가리키는 대체 URIs 전송

  14. SIP Messages • Request Messages • (METHODS) • INVITE • ACK • BYE • CANCEL • REGISTER • OPTION • Response Messages • (STATUS CODE) • 1xx : Informational • 2xx : Success • 3xx : Redirection • 4xx : Client Error • 5xx : Server Error • 6xx : Global Error

  15. SIP Request Messages

  16. SIP Message Syntax : Request Start line INVITE sip:bob@example.com SIP/2.0 To: Bob <sip:bob@biloxi.com> From: sip:alice@atlanta.com;tag=4711 Subject : Congratulations! Content-Length : 177 Content-Type : application/sdp Call-ID : af1234@pc33.atlanta.com CSeq : 1 INVITE Max-Forward : 70 Contact : sip:alice@pc33.atlanta.com:5066;transport=udp Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776as Message headers v=0 o=alice 2345566342 2346553445 IN IP4 pc33.atlanta.com s= c=IN IP4 pc33.atlanta.com t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Message body (SDP content)

  17. SIP Response Messages

  18. SIP Response(1/3) • 300 Multiple Choices • 301 Moved Permanently • 302 Moved Temporarily • 305 Use Proxy • 380 Alternative Service • 100 Trying • 180 Ringing • 181 Call Is Being Forwarded • 182 Queued • 183 Session Progress • 200 OK

  19. SIP Response(2/3) • 414 Request-URI Too Long • 415 Unsupported Media Type • 416 Unsupported URI Scheme • 420 Bad Extension • 421 Extension Required • 423 Interval Too Brief • 480 Temporarily Unavailable • 481 Call/Transaction Does Not Exist • 482 Loop Detected • 483 Too Many Hops • 484 Address Incomplete • 400 Bad Request • 401 Unauthorized • 402 Payment Required • 403 Forbidden • 404 Not Found • 405 Method Not Allowed • 406 Not Acceptable • 407 Proxy Authentication Required • 408 Request Timeout • 410 Gone • 413 Request Entity Too Large

  20. SIP Response(3/3) • 505 Version Not Supported . • 513 Message Too Large • 600 Busy Everywhere • 603 Decline • 604 Does Not Exist Anywhere • 606 Not Acceptable • 485 Ambiguous • 486 Busy Here • 487 Request Terminated • 488 Not Acceptable Here • 491 Request Pending • 493 Undecipherable • 500 Server Internal Error • 501 Not Implemented • 502 Bad Gateway • 503 Service Unavailable • 504 Server Time-out

  21. SIP Message Syntax : Response Start line SIP/2.0 200 OK To: Bob <sip:bob@biloxi.com>;tag=428 From: sip:alice@atlanta.com;tag=4711 Subject : Congratulations! Content-Length : 121 Content-Type : application/sdp Call-ID : af1234@pc33.atlanta.com CSeq : 1 INVITE Max-Forward : 70 Contact : sip:bob@192.0.2.4 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776as Message headers v=0 o=bob 2890844526 2890844526 IN IP4 192.0.2.4 s= c=IN IP4 192.0.2.4 t=0 0 m=audio 5000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Message body (SDP content)

  22. SIP Headers(1/2) • Date • Error-Info • Expires • From • In-Reply-To • Max-Forwards • Min-Expires • MIME-Version • Organization • Priority • Proxy-Authenticate • Proxy-Authorization • Proxy-Require • Record-Route • Accept • Accept-Encoding • Accept-Language • Alert-Info • Allow • Authentication-Info • Authorization • Call-ID • Call-Info • Contact • Content-Disposition • Content-Encoding • Content-Language • Content-Length • Content-Type • CSeq

  23. SIP Headers(2/2) • Reply-To • Require • Retry-After • Route • Server • Subject • Supported • Timestamp • To • Unsupported • User-Agent • Via • Warning • WWW-Authenticate

  24. SIP - Extensions • Basic SIP Specifications • RFC 3261 : SIP (Session Initiation Protocol) • RFC 3263 : Locating SIP Servers • RFC 3264 : An Offer/Answer Model with the SDP • SIP Extensions • METHOD Extensions • HEADER Extensions • Security and Privacy Support • SIP WG Activities(2004.8) • RFC: 23, Internet Drafts : 22 • SIPPING WG Activities(2004.8) • RFC: 12, Internet Drafts : 31

  25. SDP • Session Description Protocol (RFC2327) • IETF MMUSIC(Multiparty Multimedia Session Control) WG • Purpose • On the Mbone, to describe session information of multimedia conference • SDP Information • Session Description • Time Description • Media Description

  26. SDP Format Session Description (*) Optional Fields

  27. SDP Format Time Description Media Description . . .

  28. SIP Functional Layers User Hook on/off Ringing Session creation Application-specific processing Transaction User Transaction handling Request retransmission Transaction Layer Transport Layer Send/receive SIP message Syntax & Encoding Message parsing TLS UDP TCP SCTP Transport Protocol

  29. SIP Definitions • Call • A call is an informal term that refers to some communication between peers, generally set up for the purposes of a multimedia conversation • 각 Call들은 Call-ID 헤더로 구분 • Dialog • A dialog is a peer-to-peer SIP relationship between two UAs that persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is identified by a call identifier, local tag, and a remote tag. • 각 Dialog들은 Call-ID, From, To로 구분 • Transaction • A SIP transaction occurs between a client and a server and comprises all messages from the first request sent from the client to the server up to a final (non-1xx) response sent from the server to the client. • 각 Transaction들은 Call-ID, From, To, CSeq로 구분

  30. UAC Behavior • Generating the Request • Sending the Request • Processing Responses

  31. UAS Behavior • Method Inspection • Header Inspection • Content Processing • Content-Type, Content-Language, Content-Encoding • Applying Extensions • Processing the Request • INVITE, ACK, REGISTER, OPTIONS, BYE… • Generating the Response

  32. Registrar Behavior • Register/Update/Delete • Authentication • Challenge : WWW-Authenticate Header • Credential : Authorization Header Bob SIP Server REGISTER F1 REGISTER sips:ss2.biloxi.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob <sips:bob@biloxi.example.com>;tag=a73kszlfl To: Bob <sips:bob@biloxi.example.com> Call-ID: 1j9FpLxk3uxtm8tn@biloxi.example.com CSeq: 1 REGISTER Contact: <sips:bob@client.biloxi.example.com> Content-Length: 0 401 Unauthorized F2 REGISTER F3 200 OK F4

  33. SIP Proxy • Call Stateful Proxy • 콜이 종료될 때까지 관련 정보들을 유지 • 콜의 시작시점과 종료 시점 등에 대한 정보를 알 수 있어 과금등이 용이 • Forking 가능 • Transaction Stateful Proxy • 트랜잭션 단위로 관련 정보 유지 • Forking 가능 • Stateless Proxy • 콜에 관련된 어떠한 정보도 유지 하지 않음 • Request는 Location Server내에 유지된 주소로 전달 • Response는 Via 헤더내 명기된 주소로 전달 • 빠른 처리 속도 • Provisional Response 제공하지 않음.

  34. Proxy Behavior • Request Processing • Preprocessing Route Information • Determining Request Targets • Request Forwarding • Post-process routing information • Response Processing • Find the appropriate response context • Update timer C for provisional response • Remove the topmost Via • Add the response to the response context • Check to see if this response should be forwarded immediately • When necessary, choose the best final response from the response context • Aggregate authorization header field values if necessary • Optionally rewrite Record-Route header field calues • Forward the response • Generate any necessary CANCEL requests

  35. SIP Vision • VoIP (Voice/Video over IP) • H.323, MEGACO 등과 함께 시장을 share • SIP의 영역이 계속 확장 중 • 컨퍼런스 • IMPP (Instance Messaging & Presence Protocol) • SIP기반 인스턴스 메신저 • 홈 네트워킹 • 3GPP/3GPP2 • 3GPP/3GPP2의 기본 시그널링 프로토콜로 채택 • ITU-T NGN (Next Generation Network) • NGN의 기본시그널링 프로토콜로 채택 • OMA (Open Mobile Alliance) • SIP기반 PTT 서비스

  36. More to go… • NAT 및 방화벽 • 여러 방법들이 제시되고 있으나 아직까지 완벽한 솔루션은 제공되지 못하고 있음. • UPnP/TURN/STUN/MIDCOM/ICE • 보안 • 시그널링 보안 : S/MIME, TLS • 미디어 보안 : SRTP • SPAM • 긴급통신 • Lawful Interception • DTMF • In-band • Out-band

  37. Examples & misc.

  38. SIP Registration

  39. SIP기반 음성 통화

  40. SIP기반 음성 통화

  41. IP-PSTN 통화

  42. 상호운용성 시험 • 국내 Bake-off : ‘01.10. ’01.11. • ION 2001 : ‘01.11.25~26 • IMTC/ETSI/TTC Winter Interop! : ‘01.12.3~7, 고베 (일본) • 10th SiPit (SIP Interoperability Testing) : ‘02.4.22 ~ 26, 깐느 (프랑스) • ION 2003 : ‘03.1.13~17, TTA • 12th SiPit : ‘03.2.24~28, 스톡홀름 (스웨덴) • 15th SIPit : ’04.08.22~27, 타이페이(대만) • ION 2004 : ‘04.9.13~17, TTA

  43. 10th SIPit (Cannes, France)

  44. 12th SIPit (Stockholm, Sweden)

  45. Q & A

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