This document provides an overview of the Session Initiation Protocol (SIP) and its relevance in Voice over Internet Protocol (VoIP) technology, including its history, architecture, and various SIP messages. It discusses the advantages of SIP over other protocols like H.323 and highlights the role of VoIP in modern telecommunications. Additionally, it covers network elements, SIP addressing, supplementary services, and tools like Wireshark for analyzing SIP traffic.

1. Take a sip of SIP
Paul Sarstiuc, Romeo Vidrascu, Marius Trestioreanu
Bucharest
18 April 2016

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SIP – Session Initiation Protocol
 Content
– VoIP
– SIP History
– SIP Architecture
– Addressing SIP
– SIP Messages
– SIP Flow
– WireShark and SIP Filters
– SIP Flow – WireShark
– Supplementary Services
– SDP - WireShark

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VoIP stereotypes
 VoIP means free calls
 SIP is a collection of protocols
 SIP is better than H323
 SIP is a voice signaling protocol

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VoIP
 VoIP
– VoIP: Voice over Internet Protocol
 IP Telephony
 Internet Telephony
 Voice over Broadband (VoBB)
 Broadband Telephony
 Broadband Phone
– Alternative to PSTN
– Single infrastructure for Data, Voice and Video
– More demand for video conferences is easily satisfied with VoIP
– Cost savings on long distance calls
– Easier connectivity: customers are to be reached at multiple points
under the same “telephone number”
– Communication Services
 Voice/Video
 Fax
 Voice/Messaging Application

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VoIP
 VoIP Protocols
– SIP – Session Initiation Protocol
– H.323
– IMS – IP Multimedia Subsystem
– MGCP – Media Gateway Control Protocol
– RTP – Real-time Transport Protocol
– RTCP – Real-time Transport Control Protocol
– SDP – Session Description Protocol
– Skype Protocol (proprietary)
– TCP – Transmission Control Protocol
– UDP – User Datagram Protocol
– TLS – Transport Layer Security

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SIP OSI
 SIP vs. OSI

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History
 SIP History
– Feb. 22, 1996 draft-ietf-mmusic-scip-00; IDMS paper
– Feb. 22, 1996 draft-ietf-mmusic-sip-00
– Dec. 2, 1996 draft-ietf-mmusic-sip-01
– March 27, 1997 draft-ietf-mmusic-sip-02
– July 31, 1997 draft-ietf-mmusic-sip-03
– November 11, 1997 draft-ietf-mmusic-sip-04
– May 14, 1998 draft-ietf-mmusic-sip-05
– June 17, 1998 draft-ietf-mmusic-sip-06
– July 16, 1998 draft-ietf-mmusic-sip-07
– August 7, 1998 draft-ietf-mmusic-sip-08
– September 18, 1998 draft-ietf-mmusic-sip-09
– September 28, 1998 Last call
– November 12, 1998 draft-ietf-mmusic-sip-10
– December 15, 1998 draft-ietf-mmusic-sip-11
– January 15, 1999 draft-ietf-mmusic-sip-12
– February 2, 1999 Approved
– March 17, 1999 RFC 2543
– July 3, 2002 RFC 3261 (SIP: Session Initiation Protocol), RFC 3262 (Reliability of Provisional Responses
in Session Initiation Protocol (SIP)), RFC 3263 (Session Initiation Protocol (SIP): Locating SIP Servers), RFC
3264 (An Offer/Answer Model with Session Description Protocol (SDP)), RFC 3265 (Session Initiation Protocol
(SIP)-Specific Event Notification), RFC 3266 (Support for IPv6 in Session Description Protocol (SDP))
published

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SIP Architecture
 Network Elements:
– UA – User Agent
 UAC – User Agent Client [request]
 UAS – User Agent Server [response]
– Server Elements [RFC 3261]
 Proxy Server [phone – proxy – proxy – phone]
 Registrar [REGISTER]
 Redirect Server [3XX]
– Other Network Elements
 SBC – Session Border Controller
 Gateway

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SIP Architecture
 Network Elements

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SIP addressing
 URI – Uniform Resource Identifier
 sip:username:password@host:port
 E.g.: sip:John@sipgatedomain.org
 Secure transmission
– sips:… instead of sip:…
– TLS – Transport Layer Security

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SIP Messages
 SIP Requests:
– REGISTER
– INVITE
– ACK
– CANCEL
– BYE
– OPTIONS
– SUBSCRIBE
– REFER
– NOTIFY
 SIP Responses:
– Provisional (1xx)
– Success (2xx)
– Redirection (3xx)
– Client Error (4xx)
– Server Error (5xx)
– Global Failure (6xx)

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SIP Messages - REGISTER

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SIP Messages - INVITE

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SIP Messages - ACK

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SIP Messages - CANCEL

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SIP Messages - BYE

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SIP Messages - OPTIONS

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SIP Messages – 100 Trying

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SIP Messages – 200 Ok

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SIP Messages – 401 Unauthorized

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SIP Flow
 Direct SIP Call Between 2 UAs

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SIP Flow
 Call via Proxy

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SIP Flow
 Call via Proxy with No Answer

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SIP Flow
 Registration

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WireShark and SIP Filters
 Wireshark
– Free, open-source packet analyzer
– Network Troubleshooting and Analysis
– Software and communications protocol development
– Education.
 Popular Filters:
– sip
– sdp
– udp
– tcp
– rtp
– sip.To
– sip.to.addr

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SIP Flow - WireShark
 WireShark Capture

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SIP Flow - WireShark
 WireShark: Telephony – VoIP Calls – Flow

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Supplementary Services
– Call Hold
– Call Transfer
– Call Conference
– Call Forwarding
 on busy
 no answer
 unconditional

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Supplementary Services
 HOLD

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Supplementary Services
 HOLD

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Supplementary Services
 Transfer

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SDP - WireShark

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SDP - WireShark
 Session Description Protocol Version - 0
 Owner / Creator of the session or Owner / Creator. Identification is made by:
– Owner username. User.
– Session ID. ID of the session. Random number as a unique identifier of the session.
– Session Version. Version.
– Network Type. Tipe network. Always IN.
– Address Type. It can be IP4 (IPv4) or IP 6 (IPv6).
– Address (IP). IP Address. (200.57.7.197)
– Session Name. Name of the session.
 Connection Information:
– C = Connection Type Network (IN)
– Connection Address Type: (IP4 or IPv6)
– Connection Address: (200.57.7.197)
 Time Description, active time. (t): 0 0, start stop time = 0. [unrestricted and permanent session].
 Media Description, name and address (m): audio 40376 RTP / AVP 4 0 8 18. Type of data being transported (audio or telephone session in this
case), UDP port used (40 376), protocol used (Real Time Transport Protocol RTP / AVP Audio Video Profiles). Codecs formats:
– 8 G.711 PCMA
– 18 G.729
– 4 G.723
– 0 G.711 PCMU
 Media Attribute (a). This is a list of format codes outlined above with data from Sample rate or sampling frequency, fieldname, etc.
 Media Attribute (a). SendRecv. So send / receive.

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Q & A
Thank You !