Session Initiation Protocol SIP

1
Session Initiation Protocol (SIP)

• Chapter 5

2
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)

3
The Popularity of SIP

• Originally Developed in the MMUSIC
• by Handley, Schulzrinne, Schooler, and Rosenberg
• A separate SIP working group
• RFC 3261
• Many developers
• SIP MGCP/MEGACO
• The VoIP signaling in the future
• back-off or SIPit (SIP Interoperability Tests)
• Test products against each other
• Organized by SIP Forum

4

• The 18th SIPit event in Tokyo, Japan took place
  April 17-21, 2006, and will be hosted by JPNIC
• The 17th SIPit event in Stockholm, Sweden took
  place 2005-09-11 to 2005-09-16 and was hosted by
  Hotsip
• The 16th SIPit event in Banff, Canada took place
  2005-04-04 to 2005-04-08 and was hosted by Jasomi
  Networks
• The 15th SIPit event in Taiwan took place
  2004-08-23 to 2004-08-27 and was hosted by
  CCL/ITRI
• The 14th SIPit event in Cannes, France took place
  2004-02-08 to 2004-02-13 and was hosted by ETSI

5
SIP Basic Functionality

• User location determination of the end system to
  be used for communication
• User capabilities determination of the media and
  media parameters to be used
• User availability determination of the
  willingness of the called party to engage in
  communications
• Call setup "ringing", establishment of call
  parameters at both called and calling party
• Call handling including transfer and termination
  of calls.

6
SIP Philosophy

• Internet Standard
• - IETF - http//www.ietf.org
• Reuse Internet addressing (URLs, DNS, proxies)
• - Utilizes rich Internet feature set
• Reuse HTTP coding
• - Text based
• Makes no assumptions about underlying protocol
• - TCP, UDP, X.25, frame, ATM, etc.
• - Support of multicast

7
SIP Architecture

• A signaling protocol
• The setup, modification, and tear-down of
  multimedia sessions
• SIP SDP
• Describe the session characteristics
• Separate signaling and media streams

8
SIP Network Entities

• 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

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• Four types of servers
• Proxy servers
• Handle requests or forward requests to other
  servers
• Can be used for call forwarding

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• Redirect servers
• Map the destination address to zero or more new
  addresses
• Do not initiate any SIP requests

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• A user agent server
• Accept SIP requests and contacts the user
• The user responds ? an SIP response
• A SIP device
• E.g., an SIP-enabled telephone
• A registrar
• Accepts SIP REGISTER requests
• Indicating the user is at a particular address
• Typically combined with a proxy or redirect
  server

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SIP Call Establishment

• It is simple
• A number of interim responses

13
SIP Advantages

• Attempt to keep the signaling as simple as
  possible
• Offer a great deal of flexibility
• Various pieces of information can be included
  within the messages
• Including non-standard information
• Enable the users to make intelligent decisions
• The user has control of call handling
• No need to subscribe call features

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• Call Completion to Busy Subscriber service

15
Overview of SIP Messaging Syntax

• Text-based
• Similar to HTTP
• 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
• The success or failure of a given request
  

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• 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 media format
• SDP, Session Description Protocol
• Could include an ISDN User Part message
• Examined only at the two ends

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SIP Requests

• method SP request-URI SP SIP-version CRLF
• request-URI
• The address of the destination
• Methods
• INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER
• extensions INFO, REFER, UPDATE,
• INVITE
• Initiate a session
• Information of the calling and called parties
• The type of media
• IAM (initial address message) of ISUP
• ACK only the final response

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• BYE
• Terminate a session
• Can be issued by either the calling or called
  party
• Options
• Query a server as to its capabilities
• A particular type of media
• The response if sent an INVITE
• CANCEL
• Terminate a pending request
• E.g., an INVITE did not receive a final response

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• REGISTER
• Log in and register the address with a SIP server
• all SIP servers multicast address
  (224.0.1.75)
• Can register with multiple servers
• Can have several registrations with one server
• INFO
• RFC 2976
• Transfer information during an ongoing session
• DTMF digits
• account balance information
• midcall signaling information generated in
  another network

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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 code 200 is defined)
• 3XX Redirection
• 4XX Request Failure
• 5XX Server Failure
• 6XX Global Failure
• All responses, except for 1XX, are considered
  final
• Should be ACKed

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• 180 Ringing
• 181 Call Is Being Forwarded
• 182 Queued
• 183 Session Progress
• 200 OK
• 300 Multiple Choices
• 301 Moved Permanently
• 302 Moved Temporarily
• 305 Use Proxy
• 380 Alternative Service

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One number service
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SIP Addressing

• SIP URLs (Uniform Resource Locators)
• user_at_host
• E.g.,
• sipcollins_at_home.net
• sip3344556789_at_telco.net
• Supplement the URL
• sip3344556789_at_telco.netuserphone
• sipuserpassword_at_hostporturi-parameters?headers
  

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Message Headers

• Provide further information about the message
• information elements
• E.g.,
• Toheader in an INVITE
• The called party
• Fromheader
• The caling party
• Four main categories
• General, request, response, and entity headers
• A list in Table 5-2
• Mapping in Table 5-3

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General Headers

• Used in both requests and responses
• Basic information
• E.g., To, From, Call-ID,
• Contact
• A URL for future communication
• May be different from the From header
• Requests passed through proxies

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• Request Headers
• Apply only to SIP requests
• Addition information about the request or the
  client
• E.g.,
• Subject
• Priority, urgency of the request
• Authorization, authentication of the request
  originator
• Response Headers
• Further information about the response
• E.g.,
• Unsupported, features
• Retry-After

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• Entity Header
• Session information presented to the user
• Session description, SDP
• The RTP payload type, an address and port
• Content-Length, the length of the message body
• Content-Type, the media type of the message
• Content-Encoding, for message compression
• Content Disposition,
• Content-Language,
• Allow, used in a Request to indicate the set of
  methods supported
• Expires, the date and time

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Example of SIP Message Sequences

• Registration
• Via
• Call-ID
• host-specific
• Content-Length
• Zero, no msg body
• Cseg
• Avoid ambiguity
• Expires
• TTL
• 0, unreg
• Contact

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Invitation

• A two-party call
• Subject
• optional
• Content-Type
• application/sdp

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Via Headers and Routing

• Via headers are used for routing SIP messages
• Requests
• Request initiator puts address in Via header
• Servers check Via with senders address, then add
  own address, then forward. (if different, add
  received parameter)
• Responses
• Response initiator copies request Via headers.
• Servers check Via with own address, then forward
  to next Via address
• All Via headers are copied from request to
  response in order
• Response is sent to address in top Via header

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Receiving Subsequent Requests

• 1. If the Call-ID is new,
• - the request is for a new call, regardless
  of the values
• of the To and From header fields.
•  2. If the Call-ID exists,
• - the request is for an existing call.
• - If the To, From, Call-ID, and CSeq values
  exactly match (including tags) those of any
  requests received previously,
• - the request is a retransmission. 
  
• 3. If there was no match to the previous step,
• - To From fields compared against existing
  call leg local and remote addresses.
• - If there is a match, the CSeq in the
  request gt last CSeq
• received on that leg,
• - the request is a new transaction for an
  existing call leg.

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Reliability

• If UDP is used
• SIP client should retransmit a BYE, CANCEL,
  OPTIONS, or REGISTER request, exponential
  backoff, starting at a T1 second interval,
  doubling the interval for each packet, and
  capping off at a T2 second interval.
• Retransmit a INVITE request with an interval that
  starts at T1 seconds, exponential back off, cease
  retransmissions if a provisional or definitive
  response recvd., or once it has sent a total of 7
  request packets
• Clients using TCP do not need to retransmit
  requests

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Termination of a Call

• Cseq
• Has changed

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Redirect Servers

• An alternative address
• 302, Moved temporarily
• Another INVITE
• Same Call-ID
• Cseq

37
Proxy Servers

• Entity headers are omitted
• Changes the Req-URI
• Via
• The path
• Loop detected, 482
• For a response
• The 1st Via header
• Checked
• removed

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Proxy state

• Can be either stateless or stateful
• Record-Route
• The messages and responses may not pass through
  the same proxy
• Use Contact
• A Proxy might require that it remains in the
  signaling path
• In particular, for a stateful proxy
• Insert its address into the Record-Route header
• The response includes the Record-Route header
• The Record-Route header is used in the
  subsequent requests
• The Route header the Record-Route header in
  reverse order, excluding the first proxy
• Each proxy remove the next from the Route header

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Forking Proxy

• fork requests
• A user is registered at several locations
• branchxxx

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The Session Description Protocol

• The message body
• SDP, RFC 2327
• The Structure of SDP
• Session Level Info
• Name
• The originator
• The time
• Media Level Info
• Media type
• Port number
• Transport protocol
• Media format

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• SDP session description structure

44
SDP Syntax

• A number of lines of text
• In each line
• fieldvalue
• Session-level fields first
• Media-level fields
• Begin with media description field (m)

45
Mandatory Fields

• v(protocol version)
• o(session origin or creator and session id)
• s(session name), a text string
• t(time of the session)
• tltstart timegt ltstop timegt
• NTP time values in seconds
• m(media)
• mltmediagt ltportgt lttransportgt ltfmt listgt
• Media type
• The transport port
• The transport protocol
• The media format, an RTP payload format

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Optional Fileds

• i(session information)
• A text description
• At both session and media levels
• u(URI of description)
• Where further session information can be obtained
• Only at session level
• e(e-mail address)
• Who is responsible for the session
• Only at the session level
• p(phone number)
• Only at the session level

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• c(connection information)
• Connection type, network type, and connection
  address
• At session or media level
• b(bandwidth information)
• In kilobits per second
• At session or media level
• rltrepeat intervalgt ltactive durationgt ltlist of
  offsets from start- timegt
• For regularly scheduled session
• How often and how many times

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• z(timezone adjustments)
• zltadjustment timegt ltoffsetgt ltadjustment timegt
  ltoffsetgt ....
• For regularly scheduled session
• Standard time and Daylight Savings Time
• k(encryption key)
• kltmethodgtltencryption keygt
• An encryption key or a mechanism to obtain it
• At session or media level
• a(attributes)
• Describe additional attributes

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Ordering of Fields

• Session Level
• Protocol version (v)
• Origin (o)
• Session name (s)
• Session information (i)
• URI (u)
• E-mail address (e)
• Phone number (p)
• Connection info (c)
• Bandwidth info (b)
• Time description (t)
• Repeat info (r)
• Time zone adjustments (z)
• Encryption key (k)
• Attributes (a)

• Media level
• Media description (m)
• Media info (i)
• Connection info (c)
• Optional if specified at the session level
• Bandwidth info (b)
• Encryption key (k)
• Attributes (a)

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Subfields

• Field ltvalue of subfield1gt ltvalue of subfield2gt
  ltvalue of subfield3gt
• Origin (o)
• Username, the originators login id or -
• session ID
• A unique ID
• Make use of NTP timestamp
• version, a version number for this particular
  session
• network type
• A text string IN refers to Internet
• address type
• IP4, IP6
• Address, a fully-qualified domain name or the IP
  address
• omhandley 2890844526 2890842807 IN IP4
  126.16.64.4

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• Connection Data
• The network and address at which media data are
  to be received
• Network type, address type, connection address
• cIN IP4 224.2.17.12/127
• Media Information
• Media type
• Audio, video, application, data, or control
• Port, 1024-65535
• Format
• List the various types of media
• RTP/AVP payload types
• m audio 45678 RTP/AVP 15 3 0
• G.728, GSM, G.711

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• Attributes
• Property attribute
• asendonly
• arecvonly
• value attribute
• aorientlandscape
• rtpmap attribute
• The use of dynamic payload type
• artpmapltpayload typegt ltencoding namegt/ltclock
  rategt /ltencoding parametersgt.
• mvideo 54678 RTP/AVP 98
• artpmap 98 L16/16000/2

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Usage of SDP with SIP

• SIP for the establishment of multimedia sessions
• SDP a structured language for describing the
  sessions
• The entity header

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Negotiation of Media

• Fig 5-15
• G.728 is selected
• If a mismatch
• 488 or 606
• Not Acceptable
• A Warning header
• INVITE with multiple media streams
• Unsupported should also be returned
• With a port number of zero

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• Offer/answer

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• OPTIONS Method
• Determine the capabilities of a potential called
  party

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SIP Extensions and Enhancements

• RFC 2543, March 1999
• obsoleted by RFCs 3261,3262, 3263, 3265
• Will be enhanced considerably before it becomes
  an Internet standard
• 183 session progress (RFC 3261)
• Supported header (RFC 3261)
• Require
• Supported

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183 Session-Progress Message

• The addition of a new response
• Status code 183
• To open a one-way media path
• From the called party to calling party
• convey information about the progress of the call
  that is not otherwise classified
• Tones or announcements
• ACM (address complete message) of SS7
• For SIP PSTN SIP connections
• When a temporary media stream is needed
• Note that alerting signal can be
• Status code 180 (ringing)
• The temporary media stream will be terminated
• As soon as the called user answers

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The SIP Supported Header

• The Require header
• UACs tell UASs about options that the UAC expects
  the UAS to support
• require 100rel
• may receiver 420 (Bad Extension)
• The Supported header
• enumerates all the extensions supported by the
  UAC or UAS
• Included in both requests and responses
• BYE, CANCEL, INVITE, OPTIONS and REGISTER
• Should not be included in the ACK
• 421, extension required
• The UAS needs a particular extension to process
  the request

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SIP INFO Method

• A new SIP method RFC 2976
• The transfer of information in the middle of a
  call
• DTMF digits, account-balance information,
  mid-call signaling information (from PSTN)
• A powerful, flexible tool to support new services
• e.g., the users prepaid account balance

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SIP Event Notification

• SIP-specific event notification
• be informed of some event(s)
• RFC 3265
• SUBSCRIBE
• subscribe to certain event
• Event header
• NOTIFY
• inform the user
• 200 (OK) response

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SIP for Instant Messaging

• SIMPLE - SIP for Instant Messaging and Presence
  Leveraging Extensions
• a working group
• RFC 3994, 3856
• The exchange of content between a set of
  participants in near real time
• IMs are usually grouped together into brief live
  comversations
• MESSAGE request, RFC 3994
• a message body in the form text/plain, or
  message/cpim (common presence and instant
  message) using XML

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• Doesnt establish a SIP dialog
• Can be associated with an existing SIP dialog
• Contact header is forbidden
• No Record-Route or Route header

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REFER Method

• RFC 3515
• Instruct the receiver to contact a third party
• Refer-to
• Can be interpreted as an implicit SUBSCRIBE
• The sender will be notified the result
• 202 (accepted)
• An extension
• A SIP message is tunneled within a SIP message
• Refer-by

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Reliability of Provisional Responses

• Provisional Responses
• 100 (trying), 180 (ringing), 183 (session in
  progress)
• Are not answered with an ACK
• If the messages is sent over UDP
• Unreliable
• Lost provisional response may cause problems when
  interoperating with other network
• 180, 183 ? Q931 alerting or ISUP ACM
• To drive a state machine
• E.g., a call to an unassigned number
• ACM to create a one-way path

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• RSeq
• Response seq
• 1, when retxm
• Rack
• Response ACK
• PRACK
• Prov Resp ACK
• the option tag
• 100rel
• Should not
• Apply to 100
• hop-by-hop

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UPDATE Method

• RFC 3311
• Change the media format in the early state
• re-INVITE cannot be used it changes the state
• Can be used to reserve network resources

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Summary of SIP methods

• INVITE create a session
• BYE terminates a session
• ACK acknowledges a final response for an INVITE
  request
• CANCEL cancels an INVITE request
• REGISTER binds a public SIP URI to a Contact
  address
• OPTIONS queries a server for capabilities
• SUBSCRIBE installs a subscription for a resource
• NOTIFY informs about changes in the state of the
  resource
• MESSAGE delivers an Instant Message
• REFER used for call transfer, call diversion,
  etc.
• PRACK acknowledges a provisional response for an
  INVITE request
• UPDATE changes the media description (e.g. SDP)
  in an existing session
• INFO used to transport mid-session information
• PUBLISH publication of presence information

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Integration of SIP and Resource Mang

• RFC 3312
• The signaling might take a different path from
  the media
• Assume resource-reservation mechanisms available
  (Chapter 8)
• On a per-session basis
• On an aggregate basis
• A new SIP header in the INVITE
• Resources reservation is needed
• The user should not yet be alerted
• But unrecognized header is ignored

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• Integration of Resource Management and SIP for IP
  Telephony
• A new method, PRECONDITION-MET
• The far-end phone will not ring until
• Also specifies extensions to SDP
• Can define any number of preconditions in SDP
  without revise SIP every time
• The response is sent using reliable signaling
• Once the resource is reserved
• An UPDATE request is sent
• If failed, could select a lower-bandwidth codec

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• The preconditions/requirements in the SDP
• Three status
• desired, current, and confirmed
• Resource reservation
• end-to-end (e2e), local, and remote
• Purpose
• send, recv, and sendrecv
• Strength
• mandatory, optional, none and failure

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• Examples
• maudio 4444 RTP/RTCP 0
• acurr qos e2e none
• ades qos mandatory e2e sendrecv
• acurr qos e2e send
• ades qos mandatory e2e sendrecv
• acurr qos e2e sendrecv
• ades qos mandatory e2e sendrecv

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Usage of SIP for Features/Services

• Personal mobility by registration
• Can carry MIME (Multi-Purpose Internet Mail
  Extension) content
• Text, HTML documents, an image, etc.
• SIP address is a URL
• Click-to-call applications
• Supplementary Custom Local Area Signaling Service
  (CLASS) services
• Call waiting, call forwarding, multi-party
  calling, call screening
• Proxy-controlled QoS, IN SCP, INAP, OSA

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Call Forwarding

• On busy
• 486, busy here

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Consultation Hold

• A SIP UPDATE

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Windows Messenger

• SIP-based
• Runs on Windows XP systems
• Core Features
• Presence and contact list management
• Instant Messaging
• Voice and Video
• Data collaboration and File transfer
• PC to Phone
• Administrative Policies to enable/disable
  features
• Acoustic Echo Cancellation in software

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• Microsoft .Net SIP server or enterprise SIP
  server
• Can not go point-to-point using IP addressing-
  needs proxy server
• Microsoft Live Communications Server (LCS) is
  their proxy server

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SIP/H.323 Interoperability Using the Radvision MCU
Microsoft Live Communication Server
Radvision MCU
SIP
IMfirst
H.323
Microsoft Active Directory
Internet
Windows Messenger Clients
H.323 Clients
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Interworking

• PSTN Interworking
• A SIP URL
• A network gateway
• Fig. 5-27
• SIP to PSTN call
• Fig. 5-28
• PSTN to SIP call
• PSTN SIP PSTN
• MIME media types
• For ISUP and QSIG

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Interworking with H.323

• An Internet draft
• SIP-H.323 interworking gateway

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Summary

• The future for signaling in VoIP networks
• Simple, yet flexible
• Easier to implement
• Fit well with the media gateway control protocols