563.13.2 VoIP and SIP Protocol

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563.13.2 VoIP and SIPProtocols

Presented by: Milan Lathia

VoIP Group: Milan Lathia, Nalin Pai, Zahid Anwar, Mike Tucker

University of Illinois

Spring 2006


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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


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A Communication Session

Source: Avaya


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What is SIP?

Session Initiation Protocol (SIP) - An IETFprotocol for session establishment (RFC 3261): Locate the other party

Negotiate what resources/media will be used in thesession

Initiate & terminate the session

Media is transported on RTP and codecs arere-used from other call signaling protocols such

as H.323 Leverages Internet Protocols and Addressing

SIP is highly extensible Example: Presence & event platforms


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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


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SIP in the Protocol Stack


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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:[email protected] sip:[email protected]


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SIP Call Flow

Client - originates message

Server - responds to or forwards message

200 OK

ACK

INVITE sip:[email protected]

user.company.com Bob.acme.com

SIPUser Agent

Client

SIPUser Agent

Server

BYE

200 OK

Media Stream


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SIP Signaling through Proxy

[email protected] [email protected]

1: INVITE [email protected]

2: 100/Trying

8: 180/Ringing

5: INVITE [email protected]

3: [email protected] ? 4: [email protected]

6: 100/Trying

7: 180/Ringing

10: 200/OK

9: 200/OK

11: ACK [email protected]

User Agent

Server

User Agent

Client

SIP Registrar

comp2.com


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Request MethodINVITE sip:[email protected]

Via: SIP/2.0/UDP proxy.acme.com:5060

From: UserA

To: UserB

Call-ID: [email protected]: 1 INVITE

Subject: Meeting Today

Contact: sip:[email protected]

Content-Type: application/sdp

Content-Length: 147

v=0o=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

Response StatusSIP/2.0 200 OK

Via: SIP/2.0/UDP proxy.acme.com:5060

From: UserA

To: UserB

Call-ID: [email protected]: 1 INVITE

Subject: Meeting Today

Contact: sip:[email protected]

Content-Type: application/sdp

Content-Length: 134

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

SIP Requests and Responses


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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 littleoverhead.

The Real-time Transport Control Protocol

Used to report on the performance of a particular RTPtransport session.

Delivers information such as the number of packetstransmitted and received, the round-trip delay, jitterdelay, etc. that are used to measure Quality of Service inthe IP network.

QoS Constraints Latency 150 msec maximum

Jitter 30 msec maximum

Packet Loss 1% maximum

Media in SIP Session


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VoIP RTP Media Packets

8 kbps data 26.4 kbps with headers

G.729 Data PacketRTP

UDP

IP

(20) (8) (12) (20 bytes)

Type Bit-rate

kbps

Coding

Delay

Quality

(MOS)

Quality

G.711 PCM 64


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References

Ono, K., Tachimoto, S., Requirements for End-to-Middle Security for theSession 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, InternetTelephony, 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), Masters 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 Secureand Reliable VoIP Infrastructure, CEC Project No. COOP-005892, April 30,

2005.


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Questions

Team Member (Mike Tucker) Present

Newsgroup

Email: [email protected]


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Next

Overview of SIP and VoIP Security Issuesand Project Details

on April 28th, 2006

by Zahid Anwar and Mike Tucker

Final Presentation

on May 5, 2006

by Entire Team

563.13.2 VoIP and SIP Protocol

Documents

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