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Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) is a signaling protocol for setting up, modifying, and tearing down multimedia communication sessions over the Internet. It is more flexible and simpler to implement than H.323. SIP allows for advanced features and is better suited to support intelligent user devices. It is part of the IETF multimedia data and control architecture, along with protocols like SDP, RTSP, and SAP. SIP has gained popularity since its original development and is used with other protocols like MGCP/MEGACO for VoIP signaling.

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0% found this document useful (0 votes)
107 views

Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) is a signaling protocol for setting up, modifying, and tearing down multimedia communication sessions over the Internet. It is more flexible and simpler to implement than H.323. SIP allows for advanced features and is better suited to support intelligent user devices. It is part of the IETF multimedia data and control architecture, along with protocols like SDP, RTSP, and SAP. SIP has gained popularity since its original development and is used with other protocols like MGCP/MEGACO for VoIP signaling.

Uploaded by

Oshdi Alazazi
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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You are on page 1/ 23

Session Initiation Protocol (SIP)

Introduction

„ A powerful alternative to H.323


„ More flexible, simpler
„ Easier to implement
„ Advanced features
„ Better suited to the support of intelligent user
devices
„ A part of IETF multimedia data and control
architecture
„ SDP, RTSP (Real-Time Streaming Protocol), SAP
(Session Announcement Protocol)

IP Telephony 2
Popularity of SIP
„ Originally Developed in the MMUSIC (Multiparty
Multimedia Session Control)
„ A separate SIP working group
„ RFC 2543
„ Many developers
„ The latest version: RFC 3261 (June 2002 )
„ SIP + MGCP/MEGACO
„ The VoIP signaling in the future
„ “bake-offs” or SIP Interoperability Tests
„ The development of SIP and its implementation by system
developers has involved a number of events.
„ Various vendors come together and test their products against
each other
„ to ensure that they have implemented the specification correctly
„ to ensure compatibility with other implementations
IP Telephony 3
SIP Architecture
„ A signaling protocol
„ The setup, modification, and tear-down of multimedia sessions
„ SIP + SDP
„ Describe the session characteristics to potential session
participants
„ Separate signaling and media streams
„ Signaling may pass via one or more proxy or redirect servers
„ Media stream takes a more direct path.

SIP Signaling

IP Network

RTP Media Stream

SIP User SIP User


IP Telephony 4
SIP Network Entities [1/4]

„ Clients
„ User agent clients
„ Application programs sending SIP requests
„ Servers
„ Responds to clients’ requests
„ Clients and servers may be in the same
platform.
„ Proxy acts as both clients and servers

IP Telephony 5
SIP Network Entities [2/4]

„ Four types of servers


„ Proxy servers
„ Act in a similar way to a proxy server used for web
access
„ Handle requests or forward requests to other servers
after some translation
„ Can be used for call forwarding, time-of-day routing,
or follow-me services

1.Request 2.Request
Collins@work.com Collins@home.net

4.Response SIP 3.Response


Caller@work.com Proxy Collins@home.net

IP Telephony 6
SIP Network Entities [3/4]
„ Redirect servers
„ Accept SIP requests
„ Map the destination address to zero or more new
addresses
„ Return the new address(es) to the originator of the
request
1.Request
Collins@work.com

2.Moved temporarily
Contact: Collins@home.net

Caller@work.com 3.ACK Redirect Server

4.Request
Collins@home.net

5.Response
Collins@home.net IP Telephony 7
SIP Network Entities [4/4]
„ A user agent server
„ Accepts SIP requests and contacts the user
„ The user responds → an SIP response
„ A SIP device
„ E.g., a SIP-enabled telephone
„ A registrar (location server)
„ Accepts SIP REGISTER requests
„ Indicating that the user is at a particular address
„ Personal mobility
„ Typically combined with a proxy or redirect server

IP Telephony 8
SIP Call Establishment
„ SIP call establishment is simple.
„ A number of interim responses may be made to the
INVITE prior to the called party accepting the call.

INVITE
a
Ringing
b
OK
c
ACK
d
Conversation
e

BYE
f
OK
g
IP Telephony 9
SIP Advantages

„ Attempt to keep the signaling as simple as possible


„ Offer a great deal of flexibility
„ Does not care what type of media is to be exchanged
during a session or the type of transport to be used for
the media
„ Various pieces of information can be included
within the messages
„ Easily include non-standard information
„ Text-based encoding
„ Enable the users to make intelligent decisions
„ The control of the intelligent features is placed in the hands
of the customer, not the network operator.
„ E.g., SUBJECT header

IP Telephony 10
Call Completion to Busy Subscriber Service

INVITE
a
Busy (Try at 4pm)
b
ACK
c
d INVITE
Ringing
e
OK
f
ACK
g
Conversation
h

BYE
i
OK
j
IP Telephony 11
Overview of SIP Messaging Syntax
„ Text-based
„ Similar to HTTP
„ Disadvantage – more bandwidth consumption
„ SIP messages
„ message = start-line
*message-header CRLF
[message-body]
„ start-line = request-line | status-line
„ Request-line specifies the type of request
„ The response line indicates the success or
failure of a given request.

IP Telephony 12
„ Message headers
„ Additional information of the request or response
„ E.g.,
„ The originator and recipient
„ Retry-after header
„ Subject header
„ Message body
„ Describe the type of session
„ The most common structure for the message body
is SDP (Session Description Protocol).
„ Could include an ISDN User Part message
„ Examined only at the two ends

IP Telephony 13
SIP Requests [1/2]

„ Method SP Request-URI SP SIP-version CRLF


„ Request-URI
„ The SIP address of the destination
„ Methods
„ INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER
„ INVITE
„ Initiate a session
„ Information of the calling and called parties
„ The type of media
„ ~IAM (initial address message) of ISUP
„ ACK only when receiving the final response

IP Telephony 14
SIP Requests [2/2]
„ BYE
„ Terminate a session
„ Can be issued by either the calling or called party
„ OPTIONS
„ Query a server as to its capabilities
„ To support a particular type of media
„ CANCEL
„ Terminate a pending request
„ Pending Request: an INVITE did not receive a final response
„ REGISTER
„ Log in and register the address with a SIP server
„ “all SIP servers” – multicast address (224.0.1.175)
„ Can register with multiple servers
„ Can have several registrations with one server

IP Telephony 15
“One Number” Service

IP Telephony 16
SIP INFO Method

„ Specified in RFC 2976


„ For transferring information during an ongoing
session
„ The transfer of DTMF digits
„ The transfer of account balance information
„ Pre-paid service
„ The transfer of mid-call signaling information

IP Telephony 17
SIP Responses
„ SIP Version SP Status Code SP Reason-Phrase CRLF
„ Reason-Phrase
„ A textual description of the outcome
„ Could be presented to the user
„ Status code
„ A three-digit number
„ 1XX Informational
„ 2XX Success (only 200 is defined: the request has been understood and
has been performed)
„ 3XX Redirection (302: the called party is not available at the address used
in the request, and the request should be reissued to a new address
included with the response)
„ 4XX Request Failure (401: the client is not authorized to make the request)
„ 5XX Server Failure (505: the server does not support the SIP version
specified in the request)
„ 6XX Global Failure (600: busy)
„ All responses, except for 1XX, are considered final
„ Should be ACKed if the original message happened to be an INVITE
IP Telephony 18
SIP Addressing

„ SIP URIs (Uniform Resource Indicators)


„ user@host
„ sip:collins@home.net
„ sip:3344556789@telco.net

IP Telephony 19
Message Headers

„ Provide further information about the message


„ E.g.,
„ To:header in an INVITE
„ The called party
„ From:header
„ The calling party
„ Four main categories
„ General, Request, Response, and Entity headers

IP Telephony 20
General Headers
„ Used in both requests and responses
„ Basic information
„ E.g., To:, From:, Call-ID: (uniquely identifies a specific
invitation to a session), …
„ Contact:
„ Provides a URL for use in future communication regarding a
particular session
„ Examples 1: In a SIP INVITE, the Contact header might be
different from the From header.
„ An third-party administrator initiates a multiparty session.
„ Example 2: Used in response, it is useful for directing
further requests directly to the called user.
„ Example 3: It is used to indicate a more appropriate
address if an INVITE issued to a given URI failed to reach
the user.
IP Telephony 21
„ Request Headers
„ Apply only to SIP requests
„ Addition information about the request or the client
„ E.g.,
„ Subject:
„ Priority: urgency of the request (emergency, urgent,
normal, or non-urgent)
„ Response Headers
„ Further information about the response that cannot
be included in the status line
„ E.g.,
„ Unsupported
„ Retry-After

IP Telephony 22
Entity Headers
„ Indicate the type and format of information
included in the message body
„ Content-Length: the length of the message
body
„ Content-Type: the media type of the message
body
„ E.g., application/sdp
„ Content-Encoding: for message compression
„ Content Disposition: how a message part
should be interpreted
„ session, alert, render …

IP Telephony 23

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