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 SIP: Protocol (majority of presentation) VoIP: Protocol

Administration
Questions Next Steps

A Communication Session

Source: Avaya
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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
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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. 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) Focuses on the application of SIP to instant messaging and presence Does not count individual drafts likely to be promoted to WG status

Session Initiation Protocol Project INvestiGation (SIPPING)

SIP Instant Messaging and Presence Leveraging Extensions (SIMPLE)

Currently, 14 SIP + 31 SIPPING + 19 SIMPLE WG Internet Drafts = 64 total

SIPit and SIMPLEt Interoperability Events (SIP Forum)

Held every 6 months 15th instance just completed

Codec Standards (G.711, G.723.1, H.264,) Standards (H.323, H.320,)

International Telecommunication Union (ITU) ETSI, IMTC

Interoperability, inter-working & standards

SIP in the Protocol Stack

SIP Entities
SIP Registrar Registration
Resolution

Signaling

SIP Proxy
100.101.102.103

SIP Proxy Media

SIP User Agent (Client)

sip:bob@abc.com

SIP User Agent (Server)

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SIP Trapezoid
Proxy Proxy

Hop 2
Hop 1
CompanyA.com CompanyB.com

Hop 3

Media Stream Direct Path

sip:mike@CompanyA.com

sip:bob@CompanyB.com

Session Management (TCP/UDP) Media (RTP over UDP)

SIP Call Flow

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

SIP Signaling through Proxy

SIP Registrar

3: fred@comp2.com ?

4: fred@10.1.1.8

User Agent Client

1: INVITE fred@comp2.com 2: 100/Trying 5: INVITE fred@10.1.1.8

User Agent Server comp2.com

6: 100/Trying

mike@comp1.com

8: 180/Ringing

7: 180/Ringing

fred@comp2.com

9: 200/OK 10: 200/OK

11: ACK fred@hr.comp2.com

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SIP Requests and Responses

Request Method
INVITE sip:UserA@acme.com
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
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

Response Status
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:UserB@100.111.112.113 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

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Media in SIP Session

Real-time Transport Protocol (RTP) is used to transport realtime 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

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VoIP RTP Media Packets

Type
G.711 G.726 PCM ADPCM

Bit-rate kbps
64 32

Coding Delay
<1ms <1ms

Quality (MOS)
4.2 4.0

Quality
Good Good

G.728
GSM G.729 G.723.1 I P

CELP
RPE-LTP CELP CELP U D P R T P

16
13.2 8 6.4

2 ms
2ms 5ms 7.5

4.0
3.7 4.0 3.8

Good
Fair-Good Good Fair-Good

G.729 Data Packet (20 bytes)

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(20)

(8) (12)

8 kbps data 26.4 kbps with headers

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-ietfidentity-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 Secure and Reliable VoIP Infrastructure, CEC Project No. COOP-005892, April 30, 2005.

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

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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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