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RFC 3261 Session Initiation Protocol (SIP)


RFC 3261+ Session Initiation Protocol (SIP) AN Lab May 19, 2006 Kim, YangJung 1996 Mark Hadley s SIP(Session Invitation Protocol) Henning Schulzrinne s SCIP ... – PowerPoint PPT presentation

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Title: RFC 3261 Session Initiation Protocol (SIP)

RFC 3261 Session Initiation Protocol (SIP)
  • AN Lab
  • May 19, 2006
  • Kim, YangJung

  • Presentation Outline
  • ? PART I ?
  • What is SIP?
  • SIP Protocol Updates
  • SIP Protocol Structure
  • SIP Users / Work groups
  • ? PART II ?
  • SIP in a similar Domain
  • SIP in a dissimilar Domain
  • Proxy servers in SIP
  • Complimentary protocols in SIP
  • Concerns about SIP
  • SIP a new generation of service

Advanced Network Laboratory 2
Kim, YangJung
  • What is SIP?
  • SIP (Session Initiation Protocol) is an
    application-layer control protocol that can
    establish, modify and terminate multimedia
    sessions such as Internet telephony calls (VOIP),
    multimedia distribution, multimedia conferences,
    chat, interactive games and virtual reality.
  • SIP is very much like HTTP, the Web protocol, or
    SMTP. Messages consist of headers and a message
  • SIP is a text-based protocol that uses UTF-8
  • SIP uses port 5060 both for UDP and TCP. SIP may
    use other transports
  • SIP does not provide services. Rather, SIP
    provides primitives that can be used to implement
    different services.

Advanced Network Laboratory 3
Kim, YangJung
  • What is SIP? (cont.)
  • SIP supports 5 facets of establishing and
    terminating multimedia communications
  • User location determination of the end system to
    be used for communication
  • User availability determination of the
    willingness of the called party to engage in
  • User capabilities determination of the media and
    media parameters to be used
  • Session setup "ringing", establishment of
    session parameters at both called and calling
  • Session management including transfer and
    termination of sessions, modifying session
    parameters, and invoking services

Advanced Network Laboratory 4
Kim, YangJung
  • SIP Protocol updates
  • 1996
  • Mark Hadleys SIP(Session Invitation Protocol)
  • Henning Schulzrinnes SCIP(Simple Conference
    Control Protocol)
  • 1999. 3.
  • RFC 2543 by IETF MMUSIC WG
  • 1999. 9.
  • 2000. 6
  • RFC 2543bis
  • 2002. 6
  • RFC 3261

Advanced Network Laboratory 5
Kim, YangJung
  • SIP Protocol updates
  • RFC2543 Session Initiation Protocol Obsolete
    It is also an application-layer control
    (signaling) protocol for creating, modifying and
    terminating sessions with one or more
    participants. These sessions include Internet
    multimedia conferences, Internet telephone calls
    and multimedia distribution. Members in a session
    can communicate via multicast or via a mesh of
    unicast relations, or a combination of these.
  • RFC3261 Session Initiation Protocol (SIP)
  • RFC3262 SIP Reliable provisional response
    messaging It is an extension to the SIP
    providing reliable provisional response messages.
    This extension uses the option tag 100rel and
    defines the Provisional Response ACKnowledgement
    (PRACK) method.
  • RFC3263 Locating SIP server It uses DNS
    procedures to allow a client to resolve a SIP
    Uniform Resource Identifier (URI) into the IP
    address, port, and transport protocol of the
    next hop to contact. It also uses DNS to allow a
    server to send a response to a backup client if
    the primary client has failed.

Advanced Network Laboratory 6
Kim, YangJung
  • SIP Protocol updates (cont.)
  • RFC3264 An Offer/Answer Model with the Session
    Description Protocol (SDP) In the model, one
    participant offers the other a description of the
    desired session from their perspective, and the
    other participant answers with the desired
    session from their perspective. This
    offer/answer model is most useful in unicast
    sessions where information from both participants
    is needed for the complete view of the session.
    The offer/answer model is used by protocols like
    the Session Initiation Protocol (SIP).
  • RFC3265 Session Initiation Protocol
    (SIP)-Specific Event Notification It is an
    extension to the SIP. The purpose of this
    extension is to provide an extensible framework
    by which SIP nodes can request notification from
    remote nodes indicating that certain events have
  • We need to note that RFC3265 is NOT intended to
    be a general-purpose infrastructure for all
    classes of event subscription and notification.

Advanced Network Laboratory 7
Kim, YangJung
  • SIP Protocol structure
  • SIP is structured as a layered protocol.
  • Each layer is independent but loosely coupled
  • Lower layer is encoded in BNF
  • Transport layer defines how client and server
    send and receives responses
  • Transaction layer handles application-layer
    retransmissions, matching of responses to
    requests, and application-layer timeouts.
  • Transaction users comprise all SIP entities
    except stateless proxies.

Transaction User (TU)
Syntax and Encoding
Advanced Network Laboratory 8
Kim, YangJung
  • SIP Protocol structure
  • SIP protocol defines several methods
  • Methods in SIP RFC
  • INVITE When a user agent client desires to
    initiate a session (for example, audio, video, or
    a game), it formulates an INVITE request and
    sends it to one or more user agent server (UAS).
  • re-INVITE An INVITE request sent within an
    existing dialog is known as a re-INVITE to
    changing addresses or ports, adding a media
    stream, deleting a media stream, and so on.
  • REGISTER Registration entails sending a REGISTER
    request to a special type of UAS known as a
    Registrar/Location server.
  • ACK Used to facilitate reliable message exchange
    for INVITEs.
  • CANCEL Used to cancel an invitation.
  • BYE The BYE request is used to terminate a
    specific session or attempted session.
  • OPTIONS The SIP method OPTIONS allows a UA to
    query another UA or a proxy server as to its
    capabilities. This allows a client to discover
    information about the supported methods, content
    types, extensions, codecs, etc. without "ringing"
    the other party.

Advanced Network Laboratory 9
Kim, YangJung
  • SIP Protocol structure (cont.)
  • Methods extensions from other RFCs
  • SIP method info Extension in RFC 2976
  • SIP method notify Extension in RFC 2848 PINT
  • SIP method subscribe Extension in RFC 2848 PINT
  • SIP method unsubscribe Extension in RFC 2848
  • SIP method update Extension in RFC 3311
  • SIP method message Extension in RFC 3428
  • SIP method refer Extension in RFC 3515
  • SIP method prack Extension in RFC 3262
  • SIP Specific Event Notification Extension in RFC
  • SIP Message Waiting Indication Extension in RFC
  • SIP method PUBLISH Extension is RFC 3903

Advanced Network Laboratory 10
Kim, YangJung
  • SIP Protocol structure (cont.)
  • SIP responses codes
  • 1xx Provisional request received, continuing
    to process the request
  • 2xx Success the action was successfully
    received, understood, and accepted
  • 3xx Redirection further action needs to be
    taken in order to complete the request
  • 4xx Client Error the request contains bad
    syntax or cannot be fulfilled at this server
  • 5xx Server Error the server failed to fulfill
    an apparently valid request
  • 6xx Global Failure the request cannot be
    fulfilled at any server.

Advanced Network Laboratory 11
Kim, YangJung
  • SIP Protocol structure (cont.)
  • SIP responses example with SDP

Advanced Network Laboratory 12
Kim, YangJung
  • SIP Architecture
  • Fundamental Message Procedures

SIP Redirect Server
Location Service
Location Server
SIP Proxy
SIP Client (UACUser Agent Client)
SIP Proxy
SIP Client (User Agent Server)
Advanced Network Laboratory 13
Kim, YangJung
  • SIP Architecture (E-to-E)
  • End-to-End Scenario

Advanced Network Laboratory 14
Kim, YangJung
  • SIP Users / Workgroups
  • IP Telephony (VoIP and beyond)
  • AVT - Audio/Video Transport (RTP)
  • http//www.ietf.org/html.charters/avt-charter.html
  • IPTEL- IP Telephony (CPL, GW location)
  • SIP- signaling for call setup
  • MMUSIC Multiparty Multimedia Session Control
  • (SIP, SDP, conferencing)
  • SIPPING- Session Initiation Proposal
  • SIP
  • Vonage

Advanced Network Laboratory 15
Kim, YangJung
  • SIP Users / Workgroups (cont.)
  • Instant Messaging and Presence
  • IMPP - Instant Messaging and Presence Protocol
  • SIMPLE - SIP for Instant Messaging and Presence
  • Leveraging Extensions
  • XMPP - open, XML-based protocol for near
  • extensible messaging and presence
  • SIP typically is used over UDP or TCP,
  • it could, without technical changes, be run
  • over IPX, or carrier pigeons, frame relay,
  • ATM AAL5 or X.25, in rough order of
  • desirability.

Advanced Network Laboratory 16
Kim, YangJung
  • SIP
  • ? PART II ?

Advanced Network Laboratory 17
Kim, YangJung
  • SIP Operation

Advanced Network Laboratory 18
Kim, YangJung
  • SIP in a similar Domain

1. Call USER B
2. Query Where is USER B?
3. Response USER B SIP address
4. Proxied Call
5. Response
6. Response
7. Multimedia Channel established
Advanced Network Laboratory 19
Kim, YangJung
  • SIP in a dissimilar Domain

  • Advanced Network Laboratory 20
    Kim, YangJung

  • SIP in a dissimilar Domain
  • Call USER B
  • 2. Query How to get to USER B, DOMAIN B?
  • 3. Response Address of Proxy controller for
  • 4. Call Proxied to SIP Proxy for Domain B.
  • 5. Query Where is USER B?
  • 6. USER Bs address
  • 7. Proxied call
  • 8. Response
  • 9. Response
  • 10. Response
  • 11. Multimedia Channel established

Advanced Network Laboratory 21
Kim, YangJung
  • Proxy Servers in SIP
  • SIP is a HTTP-like, textual, client server
    protocol, using email like address.
  • Proxy servers take care of setting up sessions
    betweens users.
  • Signals and media takes different path

DNS SRV Query? iptel.org Reply IP Address of
iptel.org SIP server
INVITE sipmike_at_195.37.78.173 From
sipCaller_at_sip.comtag12 To sip
mike_at_iptel.org Call-ID 345678_at_sip.com
INVITE sipmike_at_iptel.org From
sipCaller_at_sip.comtag12 To sip
mike_at_iptel.org Call-ID 345678_at_sip.com
OK 200 From sipCaller_at_sip.comtag12 To sip
mike_at_iptel.org tag 34 Call-ID 345678_at_sip.com
OK 200 From sipCaller_at_sip.comtag12 To sip
mike_at_iptel.org tag 34 Call-ID 345678_at_sip.com
Media Stream
Advanced Network Laboratory 22
Kim, YangJung
  • Proxy Servers in SIP (cont.)
  • Proxy servers maintain central role in SIP
  • They glue SIP components such as phones,
    gateways, applications and other domains.
  • They provide place for service implementation
    (missed calls, forwarding, screening, etc.) and
    service access control

SMS Gateway
PSTN Gateway
IP Phone pool
Other Domains
Advanced Network Laboratory 23
Kim, YangJung
  • Complimentary protocols in SIP
  • SIP is not a vertically integrated communications
    system. SIP is rather a component that can be
    used with other IETF protocols to build a
    complete multimedia architecture. Typically,
    these architectures will include protocols are
  • Real-time Transport Protocol (RTP) (RFC 1889)
    for transporting real-time data and providing QoS
  • Real-Time streaming protocol (RTSP) (RFC 2326)
    for controlling delivery of streaming media.
  • Media Gateway Control Protocol (MEGACO) (RFC
    3015) for controlling gateways to the Public
    Switched Telephone Network (PSTN) and
  • Session Description Protocol (SDP) (RFC 2327)
    for describing multimedia sessions.
  • Therefore, SIP should be used in conjunction with
    other protocols in order to provide complete
    services to the users. However, the basic
    functionality and operation of SIP does not
    depend on any of these protocols.

Advanced Network Laboratory 24
Kim, YangJung
  • Comparison SIP and H.323

SIP H.323
speed High (simplicity , use UDP) Low (complexity, use TCP)
multicast Yes No
URL usage URL itself (H.225)URL in H.323protocol
Call prioritization The priority header field Overlook
Encoding Text encoding Binary encoding(ASN.1)
Advanced Network Laboratory 25
Kim, YangJung
  • Concerns about SIP
  • NAT Traversal SIP will make networks more
    vulnerable, the source may be the firewall and
    NAT (Network Address Translator) issue. SIP
    communication doesn't traverse most installed
    firewalls and NATs.
  • Built-in delay VOIP/SIP by it's nature has some
    built in delay. It digitizes small blocks of your
    conversation, compresses it, sends it to a PBX in
    packet form which intern converts it from one
    format to another if needed, then sends the
    packets on to their destination which decompress
    and converts it back into voice.

Advanced Network Laboratory 26
Kim, YangJung
  • SIP a new generation of service
  • Virtually eliminates long distance cost
  • More features available in VoIP than
    conventional phones.
  • For businesses, VoIP is a great way to manage
    operating costs in the office giving you
    economical, reliable, scalable, non pro priority
    phone system options.
  • SIP makes it easier to develop and debug
    applications leading to lower product costs for
    equipment providers.
  • SIP a new generation of service to the future .!

Advanced Network Laboratory 27
Kim, YangJung
  • References
  • Iptel.org http//www.iptel.org
  • RFC search engine http//www.rfc-editor.org/cgi-b
  • SIP http//www.cs.columbia.edu/sip/overview.html
  • SIP Knowledge http//www.sipknowledge.com/SIP_RFC
  • VoIP Info. http//www.voip-info.org/wiki-SIP

Advanced Network Laboratory 28
Kim, YangJung
  • Q A Session
  • ? Any Questions? ?
  • ? Any Questions? ?

Advanced Network Laboratory 29
Kim, YangJung
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