563.13.2 VoIP and SIP Protocols doc

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 – 15 th 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 SIP Proxy SIP Proxy Media SIP User Agent (Client) SIP User Agent (Server) Signaling Registration Resolution sip:bob@abc.com 100.101.102.103 8 SIP Trapezoid CompanyA.com CompanyB.com Hop 1 Hop 3 Hop 2 Media Stream – Direct Path Proxy Proxy Session Management (TCP/UDP) Media (RTP over UDP) sip:mike@CompanyA.com sip:bob@CompanyB.com 9 SIP Call Flow • Client - originates message • Server - responds to or forwards message 200 OK ACK INVITE sip:bob@acme.com user.company.com Bob.acme.com SIP User Agent Client SIP User Agent Server BYE 200 OK Media Stream 10 SIP Signaling through Proxy mike@comp1.com fred@comp2.com 1: INVITE fred@comp2.com 2: 100/Trying 8: 180/Ringing 5: INVITE fred@10.1.1.8 3: fred@comp2.com ? 4: fred@10.1.1.8 6: 100/Trying 7: 180/Ringing 10: 200/OK 9: 200/OK 11: ACK fred@hr.comp2.com User Agent Server User Agent Client SIP Registrar comp2.com [...].. .SIP Requests and Responses Request Method INVITE sip: UserA@acme.com Response Status SIP/ 2.0 200 OK Via: SIP/ 2.0/UDP proxy.acme.com:5060 From: UserA To: UserB 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... Infrastructure”, CEC Project No COOP-005892, April 30, 2005 14 Questions • Team Member (Mike Tucker) Present • Newsgroup • Email: milan1@uiuc.edu 15 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 16 ... 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... 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 14 Questions • Team... 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 To: UserB 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... 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 12 VoIP RTP Media Packets Type Bit-rate kbps G.711 PCM 64 . G. 723 .1, H .26 4,…) – Standards (H. 323 , H. 320 ,…) • ETSI, IMTC – Interoperability, inter-working & standards 6 SIP in the Protocol Stack 7 SIP Entities SIP Registrar SIP Proxy SIP Proxy Media SIP User. 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 20 06 2 Agenda [We start. (MOS) Quality G.711 PCM 64 <1ms 4 .2 Good G. 726 ADPCM 32 <1ms 4.0 Good G. 728 CELP 16 2 ms 4.0 Good GSM RPE-LTP 13. 2 2ms 3.7 Fair-Good G. 729 CELP 8 5ms 4.0 Good G. 723 .1 CELP 6.4 7.5 3.8 Fair-Good 14 References • Ono,