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563.13.2 VoIP and SIP Protocols

563.13.2 VoIP and SIP Protocols. Presented by: Milan Lathia VoIP Group: Milan Lathia, Nalin Pai, Zahid Anwar, Mike Tucker University of Illinois Spring 2006. Agenda. [We start where Nalin ends] Description of the VoIP & SIP Protocol A Communication Session

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563.13.2 VoIP and SIP Protocols

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  1. 563.13.2 VoIP and SIP Protocols Presented by: Milan Lathia VoIP Group: Milan Lathia, Nalin Pai, Zahid Anwar, Mike Tucker University of Illinois Spring 2006

  2. Agenda [We start where Nalin ends] Description of the VoIP & SIP Protocol • A Communication Session • SIP: Protocol (majority of presentation) • VoIP: Protocol Administration • Questions • Next Steps

  3. A Communication Session Source: Avaya

  4. What is SIP? • Session Initiation Protocol (SIP) - An IETF protocol for session establishment (RFC 3261): • Locate the other party • Negotiate what resources/media will be used in the session • Initiate & terminate the session • Media is transported on RTP and codecs are re-used from other call signaling protocols such as H.323 • Leverages Internet Protocols and Addressing • SIP is highly extensible • Example: Presence & event platforms

  5. The SIP World • IETF Working Groups involved in SIP • SIP Working Group • Maintain and continue the development of SIP and its family of extensions. • Session Initiation Protocol Project INvestiGation (SIPPING) • Document the use of SIP for applications related to telephony and multimedia, and to develop requirements for extensions to SIP needed for those applications. • Call flow examples for basic (RFC 3665), telephony (RFC 3666) and services (draft) • SIP Instant Messaging and Presence Leveraging Extensions (SIMPLE) • Focuses on the application of SIP to instant messaging and presence • Currently, 14 SIP + 31 SIPPING + 19 SIMPLE WG Internet Drafts = 64 total • Does not count individual drafts likely to be “promoted” to WG status • SIPit and SIMPLEt Interoperability Events (SIP Forum) • Held every 6 months • 15th instance just completed • International Telecommunication Union (ITU) • Codec Standards (G.711, G.723.1, H.264,…) • Standards (H.323, H.320,…) • ETSI, IMTC • Interoperability, inter-working & standards

  6. SIP in the Protocol Stack

  7. SIP Entities SIP Registrar Registration Resolution Signaling SIP Proxy SIP Proxy 100.101.102.103 Media SIP User Agent (Client) SIP User Agent (Server) sip:bob@abc.com

  8. SIP Trapezoid Proxy Proxy Hop 2 Hop 3 Hop 1 CompanyA.com CompanyB.com Media Stream – Direct Path sip:bob@CompanyB.com sip:mike@CompanyA.com Session Management (TCP/UDP) Media (RTP over UDP)

  9. SIP Call Flow • Client - originates message • Server - responds to or forwards message SIP User Agent Server SIP User Agent Client INVITE sip:bob@acme.com 200 OK ACK Media Stream BYE 200 OK user.company.com Bob.acme.com

  10. SIP Signaling through Proxy SIP Registrar 4: fred@10.1.1.8 3: fred@comp2.com ? User Agent Client User Agent Server 1: INVITE fred@comp2.com 5: INVITE fred@10.1.1.8 2: 100/Trying comp2.com 6: 100/Trying 7: 180/Ringing fred@comp2.com 8: 180/Ringing mike@comp1.com 9: 200/OK 10: 200/OK 11: ACK fred@hr.comp2.com

  11. SIP Requests and Responses Request Method Response Status INVITE sip:UserA@acme.com SIP/2.0 200 OK Via: SIP/2.0/UDP proxy.acme.com:5060 From: UserA <sip:UserA@acme.com> To: UserB <sip:UserB@acme.com> Call-ID: 123456000@acme.com CSeq: 1 INVITE Subject: Meeting Today Contact: sip:UserA@100.101.102.103 Content-Type: application/sdp Content-Length: 147 Via: SIP/2.0/UDP proxy.acme.com:5060 From: UserA <sip:UserA@acme.com> To: UserB <sip:UserB@acme.com> Call-ID: 123456000@acme.com CSeq: 1 INVITE Subject: Meeting Today Contact: sip:UserB@100.111.112.113 Content-Type: application/sdp Content-Length: 134 v=0 o=UserA 2890844526 IN IP4 acme.com s=Example Session SDP c=IN IP4 100.101.102.103 m=audio 49172 RTP/AVP 0 a=rtpmap 0:PCMU/8000 v=0 o=UserB 2890844527 IN IP4 acme.com s=Example Session SDP c=IN IP4 100.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap 0:PCMU/8000

  12. Media in SIP Session • Real-time Transport Protocol (RTP) is used to transport real-time data, such as voice or video. • “Unreliable” protocol built on top of the UDP protocol that does not guarantee delivery of packets, but which has little overhead. • The Real-time Transport Control Protocol • Used to report on the performance of a particular RTP transport session. • Delivers information such as the number of packets transmitted and received, the round-trip delay, jitter delay, etc. that are used to measure Quality of Service in the IP network. • QoS Constraints • Latency – 150 msec maximum • Jitter – 30 msec maximum • Packet Loss – 1% maximum

  13. VoIP RTP Media Packets 8 kbps data  26.4 kbps with headers UDP RTP IP G.729 Data Packet (20) (8) (12) (20 bytes)

  14. References • Ono, K., Tachimoto, S., “Requirements for End-to-Middle Security for the Session Initiation Protocol (SIP)”, IETF RFC 4189, October 2005. • Peterson, J., “Session Initiation Protocol (SIP) Authenticated Identity Body (AIB) Format”, IETF RFC 3893, September 2004. • Peterson, J., “The Role of SIP In Advancing A Secure IP World”, Internet Telephony, pp. 88-90, September 2005. • Peterson, J., Jennings, C., “Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP),”, IETF Draft draft-ietf-identity-04, February 16, 2005. • Qiu, Q., “Study of Digest Authentication for Session Initiation Protocol (SIP)”, Master’s Project Report, University of Ottawa, December 2003. • Sisalem, D., Ehlert, S., Geneiatakis, D., Kambourakis, G., Dagiuklas, T., Markl, J., Rokos, M., Boltron, O., Rodriquez, J., Liu, J., “Towards a Secure and Reliable VoIP Infrastructure”, CEC Project No. COOP-005892, April 30, 2005.

  15. Questions • Team Member (Mike Tucker) Present • Newsgroup • Email: milan1@uiuc.edu

  16. Next … • Overview of SIP and VoIP Security Issues and Project Details • on April 28th, 2006 • by Zahid Anwar and Mike Tucker • Final Presentation • on May 5, 2006 • by Entire Team

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