Session Initiation Protocol SIP

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Session Initiation Protocol (SIP)

• Chapter 5

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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)

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The Popularity of SIP

• Originally Developed in the MMUSIC
• A separate SIP working group
• RFC 2543
• Many developers
• SIP MGCP/MEGACO
• The VoIP signaling in the future
• back-off
• Test products against each other
• Will be hosted by ETSI

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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

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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

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SIP Advantages

• Attempt to keep the signaling as simple as
  possible
• 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

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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
• 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
• REGISTER
• Log in and register the address with a SIP server
• all SIP servers multicast address
  (224.0.1.1750)
• Can register with multiple servers
• Can have several registrations with one server

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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, RFC 2543
• 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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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

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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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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

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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 Syntax

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

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Mandatory Fields

• v(protocol version)
• o(session origin or creator and session id)
• s(session name), a text string
• t(time of the session), the start time and stop
  time
• m(media)
• 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
• r(repeat times)
• For regularly scheduled session
• How often and how many times
• z(timezone adjustments)
• For regularly scheduled seesion
• Standard time and Daylight Savings Time

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• k(encryption key)
• 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

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• Connection Data
• The network and address at which media data are
  to be received
• Network type
• Address type
• Connection address
• 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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• Options Method
• Determine the capabilities of a potential called
  party

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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

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Call Forwarding

• On busy
• 486, busy here

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Consultation Hold

• C0
• An address

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SIP Extensions and Enhancements

• RFC 2543, March 1999
• Proposed standard, March 2002
• SIP-T
• Include various extensions
• Will be enhanced considerably before it becomes
  an Internet standard

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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
• Enable in-band call progress information to be
  transmitted
• 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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SIP INFO Method

• A new SIP method
• 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

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The SIP Supported Header

• The Require header
• a client indicates to a server that the server
  must support certain features
• In responses
• 421, extension required
• The Supported header
• For server to know a clients capabilities
• Included in both requests and responses
• BYE, CANCEL, INVITE, OPTIONS and REGISTER
• Should not be included in the ACK

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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
• Should not
• Apply to 100

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Integration of SIP an Resource Mang

• The signaling might take a different path from
  the media
• Assume resource-reservation mechanisms available
  (Chapter 8)
• 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
• Being sent end-to-end
• aqos strength-tag SP direction-tag SP
  confirmation-tag
• asecure strength-tag SP direction-tag SP
  confirmation-tag
• If failed, could select a lower-bandwidth codec

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Interworking

• PSTN Interworking
• A SIP URL
• A network gateway
• Fig. 5-23
• SIP to PSTN call
• Fig. 5-24
• 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

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Reference Architecture of R00
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Support of Roaming Subscribers

• via the service platform in the Home Network
• via an external service platform

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UE Accessing IM Subsystem Services

• In the visited network

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Public and Private Identities

• One private user identity
• IMSI within a NAI
• One or more public user identities

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Roles of CSCF

• Call State Control Function
• SIP proxy servers
• Proxy CSCF
• The first contact point within the IM subsystem
• Forward the SIP register requests from the UE to
  an I-CSCF
• Forward the SIP messages from the UE to the
  S-CSCF
• CDR
• FFS
• Authorization, QoS management, security

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• Interrogating CSCF
• The contact point within an operators network
• Assign an S-CSCF to a user performing SIP Reg
• Obtain from HSS the address of the S-CSCF
• Forward SIP messages to the S-CSCF
• Forward SIP messages to the MGCF
• CDR
• FFS
• Inter-operator security

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• Serving CSCF
• Perform the session control services
• A registrar
• Interaction with Services Platforms
• On behalf of an originating endpoint
• On behalf of an destination endpoint
• CDR
• FFS
• Security issues

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Registration Information Flow
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Registration
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Call flow
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Different Kinds of CSCFs
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Interwork with PSTN
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