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CSc 461/561 Multimedia Systems Part C: 2. SIP

CSc 461/561 Multimedia Systems Part C: 2. SIP. Summary. Why SIP? What is SIP? SIP proxy mode SIP redirect mode SIP header SDP SIP and NAT More from Henning’s Tutorial. Why SIP (1) What we have so far …. Network: IP and IP multicast move packets from one host to other hosts

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CSc 461/561 Multimedia Systems Part C: 2. SIP

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  1. CSc 461/561Multimedia Systems Part C: 2. SIP

  2. Summary • Why SIP? • What is SIP? • SIP proxy mode • SIP redirect mode • SIP header • SDP • SIP and NAT • More from Henning’s Tutorial

  3. Why SIP (1) What we have so far … • Network: IP and IP multicast • move packets from one host to other hosts • Transport: RTP and RTCP • end2end media transport and control feedback • Application: what we can do now • voice over IP (VoIP): yes! • IP telephony: not yet • what’s missing?: signaling! (e.g., how to setup calls)

  4. Why SIP (2): What we need: signalling support • E.g., in telephone networks • in addition to speech path: voice • signaling path: out-of-band, packetized SS7 • E.g., in cellular systems (+mobility support) • handoff between base stations • roaming across service providers • On the Internet • SIP: session initiation protocol

  5. 2. SIP: quick fact (1) • SIP is not limited to IP telephony • SIP is the Internet’s signaling protocol • SIP offers • setup calls (or sessions) • make changes to ongoing calls • terminate calls, and more (e.g., presence) • SIP does not offer • media transport, QoS support, server control,etc

  6. 2. SIP design guidelines (2) • Client-server model • request-reply transaction • HTTP+MIME-like format • Common headers in plain text • request/response line (e.g., INVITE a@b.com SIP/2.0) • message headers (identification, routing, etc) • message body • e.g., session description (SDP)

  7. 2. SIP requests (3) • REGISTER: register user agents • INVITE: initiate calls • ACK: confirm responses • BYE: terminate or transfer calls • Other methods: • CANCEL, OPTIONS, INFO, COMET, PRACK, SUBSCRIBE, NOTIFY, REFER • SIP response: HTTP-like (e.g., SIP/2.0 200 OK)

  8. 2. SIP entities (4) • User agent (UA) • UA client (UAC) and UA server (UAS) • Proxy server • relay calls; chaining; forking • Redirect server • redirect calls • Registrar server • UA registration (UA whereabouts)

  9. REGISTER sip:b.com SIP/2.0From: sip:a@b.comTo: sip:a@b.comContact: <sip:1.2.3.4> Expires: 3600 a@1.2.3.4 SIP/2.0 200 OK 2. SIP: register (5) • Create UA name/location binding • name (URI): e.g., sip:a@b.com • location: sip@1.2.3.4 • valid for a specific time period locationserver sip:a@b.com b.comregistrar sip:1.2.3.4

  10. INVITE sip:a@1.2.3.4 SIP/2.0 From: sip:x@y.comTo: sip:a@b.comCall-ID: 5678@y.com INVITE sip:a@b.com SIP/2.0 From: sip:x@y.comTo: sip:a@b.comCall-ID: 5678@y.com ?a SIP/2.0 200 OK From: sip:x@y.comTo: sip:a@b.comCall-ID: 5678@y.com ACK sip:a@b.com SIP/2.0 media media BYE OK 3. SIP: proxy mode (1) locationserver b.comproxy

  11. a@west ?a@west ?a a@1.2.3.4 INVITEa@1.2.3.4 INVITEa@west.b.com INVITE a@b.com INVITE a@5.6.7.8 CANCEL INVITE a@1.2.3.4 OK 3. Proxy: chaining and forking (2) west.b.com b.com

  12. a@home.com ?a INVITE sip:a@b.com SIP/2.0 SIP/2.0 302 Moved temporarilyContact: a@home.com INVITE sip:a@home.com SIP/2.0 SIP/2.0 200 OK ACK sip:a@home.com SIP/2.0 media media 4. SIP: redirect mode (1) locationserver b.comredirect

  13. 5. SIP headers • From/To: caller/callee URI • Via: proxy routing • request/reply: use the same route • may be different from the media route • Call-ID: identification • unique at caller • Content-Type/Length: payload info • e.g., Content-Type: application/sdp

  14. 6. SDP • SDP: session description protocol • media type, network/transport parameters • e.g., media: media, port, protocol, format_list • m=audio 2000/2 RTP/AVP 0 98 • a=rtpmap:0 PCMU/8000 • connection: net_type, add_type, address • c=IN IP4 1.2.3.4/127/3 • Ref: http://www.ietf.org/rfc/rfc2327.txt

  15. 7. SIP and NAT • NAT: network address translation • was to deal with IPv4 address shortage • now pervasive in all networking scenarios • “directional” connectivity • Communications in SIP • UA<=>server • UA<=>UA • Problem: when UAC and UAS behind NAT

  16. 8. More Content More teaching content from Henning Schulzrinne’s tutorial: • http://www.cs.columbia.edu/~coms6181/slides/11/sip_long.pdf

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