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Introduction to SIP


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Title: Introduction to SIP

Introduction to SIP
  • School of Electronics and Information
  • Kyung Hee University
  • Choong Seon HONG
  • http//

Overview of SIP
  • Proposed standard released in 1999, current
    release in 2002
  • Session-layer control-plane signaling protocol
    with support for establishing, modifying and
    terminating one-to-one or multiparty sessions.
  • Light-weight, ASCII based generic signaling
    protocol to facilitate multimedia communications
    over IP
  • Independent of media characteristics and
    transport protocol properties
  • Work in IETF currently by SIP WG and SIPPING WG
  • Related work by MMUSIC WG, AAA WG, GEOPRIV WG,
    SIMPLE WG, Internet Telephony WG
  • 3GPP has decided to support it in IP multimedia
    services (IMS) domain.

Overview of SIP (contd)
  • SIP is complimentary to SGCP/MGCP
  • SIP Provides Session Control
  • SGCP/MGCP Provides Device Control

The Big FAQ and SIP?
  • Q You are too IP-centric, arent you?
  • A Of course, we are.
  • Internet telephony (which has Internet in its
    name) is about IP.
  • IP telephony runs on top of IP and utilizes the
    IP service model.
  • It is not about re-engineering PSTN -- PSTN is
    good enough.
  • SIP is much more similar to HTTP rather than to
    legacy signaling both in terms of service model
    and protocol design.

Problem description
  • Environment aging PSTN equipment, carrier fiber
    overcapacity and arrival of multimedia
    communications with preparation to millions of
    mobile hosts that are IP-aware. Additionally, the
    possibility of programming PSTN-services in a
    more open environment.
  • Problem how to enable gradual deployment and
    transition to avoid the one-big-leap-for-mankind
  • Approach IETF multimedia architecture and
    interoperability schemes with PSTN and 3GPP.
    Signaling is one part of this. The multimedia
    architectures capabilities are related to Quality
    of Service architecture and IP multicast.

IETF Multimedia Architecture
What Protocols Are Needed?
  • Signaling protocol to establish presence, locate
    users, set up, modify and teardown sessions
  • Media Transport Protocols for transmission of
    packetized audio/video
  • Supporting Protocols
  • Gateway Location, QoS, interdomain AAA,
  • address translation, IP, etc.

AAA Authentication, Authorization, Accounting
What Protocols Are There
  • Signaling SIP/SDP (IETF), H.323 (ITU-T)
  • Note SIP adopted by 3gpp lower production and
    operation costs reported
  • Media RTP (IETFs, adopted by ITU-T)
  • Transport UDP, TCP, (Stream Control
    Transmission Protocol RFC 2960)
  • Supporting protocols
  • DNS
  • TRIP - Telephony Routing over IP - discovery and
    exchange of IP telephony gateway routing tables
    between providers
  • RSVP - Resource Reservation Setup Protocol
  • COPS - Common Open Policy Service - protocol for
    for supporting policy control over QoS
  • Diameter - Authentication, Accounting,

SIP Signaling
  • SIP is end-to-end, client-server session
  • SIPs primarily provides presence and mobility
  • Protocol primitives Session setup, termination,
  • Arbitrary services built on top of SIP, e.g.
  • Redirect calls from unknown callers to secretary
  • Reply with a webpage if unavailable
  • Send a JPEG on invitation
  • Features
  • Textual encoding (telnet, tcpdump compatible)
  • Programmability

SIP - General Purpose Presence Protocol
  • SIP is not limited to Internet
  • SIP establishes user presence
  • SIP messages can convey arbitrary signaling
    payload session description, instant messages,
    JPEGs, any MIME types
  • Suitable for applications having a notion of
  • distributed virtual reality systems,
  • network games (Quake II/III implementations),
  • video conferencing, etc.
  • Applications may leverage SIP infrastructure
    (Call Processing, User Location, Authentication)
  • Instant Messaging and Presence
  • SIP for Appliances

Internet Multimedia
  • Real Time Protocol (RTP) media packets
  • Real Time Control Protocol (RTCP) monitor
  • Session Announcement Protocol (SAP)
  • Session Description Protocol (SDP)
  • Session Initiation Protocol (SIP)
  • Real Time Stream Protocol (RTSP) play out
  • Synchronized Multimedia Integration Language
    (SMIL) mixes audio/video with text and graphics
    August 2001
  • References Search keyword at http//www.rfc-edito
  • For SMIL - http//

Stages of IP Signaling Development
  • Precommercial stage (1980-1995)
  • Research in organizations and universities
  • IETF Audio/Video Transport (AVT) WG - RTP
  • IETF Multiparty Multimedia Session Control
  • PC-centric stage (1995-1998)
  • First commercial VoIP software, proprietary
  • Calls from multimedia PC to another multimedia PC
  • ITU May-June 1995 H.323v1
  • Most commercial applications H.323 compliant by
    the end of 1996
  • Carrier-grade stage (1998-)
  • Service provider VoIP deployment
  • First obstacle integration to PSTN signalling
    (gateway functionality)
  • Development of media gateway controller protocols
    (MGCP and Megaco/H.248)
  • Todays approach coexistence of SIP, H.323, MGCP
    and Megaco/H.248

SIP History
  • Work began in 1995 in IETF mmusic WG
  • 02/1996 draft-ietf-mmusic-sip-00 15 ASCII
    pages, one request type
  • 12/1996 -01 30 ASCII pages, 2 request types
  • 01/1999 -12 149 ASCII pages, 6 methods
  • 03/1999 RFC 2543, 153 ASCII pages, 6 methods
  • 11/1999 SIP WG formed
  • 11/2000 draft-ietf-sip-rfc2543bis-02, 171 ASCII
    pages, 6 methods
  • 12/2000 it was recognized that amount of work at
    SIP WG was becoming unmanageable 1 RFC 18 I-Ds
    on WGs agenda numerous individual submissions
  • 04/2001 proposal for splitting SIP WG into SIP
    and SIPPING announced
  • 2001 SIP implementations widely available
  • http//
  • http//

SIP End-devices
  • User Agent (user application)
  • UA Client (originates calls)
  • UA Server (listens for incoming calls)
  • both SW and HW available

SIP Components
  • SIP Proxy Server
  • relays call signaling, i.e. acts as both client
    and server
  • operates in a transactional manner, i.e., it
    keeps no session state
  • SIP Redirect Server
  • redirects callers to other servers
  • SIP Registrar
  • accept registration requests from users
  • maintains users whereabouts at a Location
    Server (like GSM HLR)

Telephony on the Internet may not be a
stand-alone business, but part of IP services
SIP/RTP Media Architecture
  • Public IP Backbone
  • Goes everywhere
  • End-to-end control
  • Consistent for all services
  • DNS mobility
  • Messaging
  • Web
  • Directory
  • Security
  • QoS
  • Media services
  • Sessions
  • Telephony

Telephone Gateway SIP client
CAS, Q.931, SS7
CAS channel associated signaling
Any other sessions
Commercial Grade IP Telephony
Assure baseline PSTN features Leverage and
Commonality of telephony with the Web/Internet
  • New services (new revenue)
  • Scalability (Web-like)
  • Baseline PSTNPBX features
  • Client user authentication
  • Accounting assured QoS
  • QoS assured signaling
  • Security assured signaling
  • Hiding of caller ID location
  • Better than PSTN features
  • New fast service creation
  • Internet (rapid) scalability
  • Mobility
  • Dynamic user preferences
  • End-to-end control
  • Service selection
  • Feature control
  • Mid-call control features
  • Pre-call
  • Mid-call

IP Communications
Complete integration of all services under full
user control
  • PSTN/PBX-like
  • POTS
  • AIN CS-1, CS-2
  • PBX Centrex
  • User has control of
  • All addressable devices
  • Caller and called party preferences
  • Better quality than 3.1 kHz
  • Web-like
  • Presence
  • Voice and text chat
  • Messaging
  • Voice, data, video
  • Multiparty
  • Conferencing
  • Education
  • Games
  • Any quality
  • Most yet to be invented

  • Development of SIP
  • IETF - Internet Engineering Task Force
  • MMUSIC - Multiparty Multimedia Session Control
    Working Group
  • SIP developed by Handley, Schulzrinne, Schooler,
    and Rosenberg
  • Submitted as Internet-Draft 7/97
  • Assigned RFC 2543 in 3/99
  • Internet Multimedia Conferencing Architecture.
  • Alternative to ITUs H.323
  • H.323 used for IP Telephony since 1994
  • Problems No new services, addressing, features
  • Concerns scalability, extensibility

  • SIP Philosophy
  • Internet Standard
  • IETF - http//
  • 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

  • SIP Clients and Servers - 1
  • SIP uses client/server architecture
  • Elements
  • SIP User Agents (SIP Phones)
  • SIP Servers (Proxy or Redirect - used to locate
    SIP users or to forward messages.)
  • Can be stateless or stateful
  • SIP Gateways
  • To PSTN for telephony interworking
  • To H.323 for IP Telephony interworking
  • Client - originates message
  • Server - responds to or forwards message

  • SIP Clients and Servers - 2
  • Logical SIP entities are
  • User Agents
  • User Agent Client (UAC) Initiates SIP requests
  • User Agent Server (UAS) Returns SIP responses
  • Network Servers
  • Registrar Accepts REGISTER requests from clients
  • Proxy Decides next hop and forwards request
  • Redirect Sends address of next hop back to
  • The different network server types may be

  • SIP Addressing
  • SIP gives you a globally reachable address
  • Callees bind to this address using SIP REGISTER
  • Callers use this address to establish real-time
    communication with callees.
  • URLs used as address data format examples
  • sipsales_at_hotel.xy geo.position48.54_-123.84_1

SIP Addressing (contd)
  • must include host, may include user name, port
    number, parameters (e.g., transport), etc.
  • may be embedded in Webpages, email signatures,
    printed on your business card, etc.
  • address space unlimited
  • non-SIP URLs can be used as well (mailto,
    http, ...)

SIP Registration
  • SIP Session Setup Example

SIP User Agent Client
SIP User Agent Server
200 OK
Media Stream
200 OK
  • Proxy Server Example
SIP Operation in Proxy Mode
Proxy Server Functionality
  • Serve as rendezvous point at which callees are
    glabally reachable
  • Perform routing function, i.e., determine to
    which hop (UA/proxy/redirect) signaling should be
  • Allow the routing function to be programmable.
    Arbitrary logic may be built on top of the
  • users signaling preferences
  • AAA
  • firewall control
  • etc.
  • Forking Several destinations may be tried for a
    request sequentially or in parallel.

Proxy Chaining
  • There may be also cases when a local outbound
    proxy may be involved
  • provides locally important call processing logic
    (e.g., identifying nearest 119)
  • manages firewall
  • provides least-gateway-cost routing service
  • IP phones must know address of the proxymay be
    configured manually or with a configuration
    protocol (DHCP, TFTP, ... )
  • In general, servers may be arbitrarily chained
  • a central companys server may distribute
    signaling to departmental servers
  • a user may want to forward incoming calls to her
    cell phone

Proxy Chaining an Example
Note signaling (in red) may take a completely
different path from media in blue)
  • Redirect Server Example

SIP Operation in Redirect Mode
SIP Server - Proxy versus Redirection
  • A SIP server may either proxy or redirect a
  • Which of the two method applies is a
    configuration issue. It may be statically
    configured or dynamically determined/
  • Redirection useful if a user moves or changes
    her provider (PSTN The number you have dialed
    is not available.) -- caller does not need to
    try the original server next time. Stateless.
  • Proxy useful if forking, AAA, firewall control
    needed. In general, proxying grants more control
    to the server.

  • SIP Requests RFC2543 Methods
  • SIP Requests (Messages) defined as
  • Method SP Request-URI SP SIP-Version CRLF
    (SPSpace, CRLFCarriage Return and Line Feed)
  • Example INVITE SIP/2.0

  • SIP Requests Example
  • Required Headers (fields)
  • Via Shows route taken by request.
  • Call-ID unique identifier generated by client.
  • CSeq Command Sequence number
  • generated by client
  • Incremented for each successive request

SIP/2.0/UDP host.wcom.com5060 From Alan
Johnston ltsipalan.johnston_at_wcom.comgt To Jean
Luc Picard ltsippicard_at_wcom.comgt Call-ID CSeq 1 INVITE

Uniquely identify this session request
  • SIP Requests Example
  • Typical SIP Request

SIP/2.0/UDP host.wcom.com5060 From Alan
Johnston ltsipalan.johnston_at_wcom.comgt To Jean
Luc Picard ltsippicard_at_wcom.comgt Call-ID CSeq 1 INVITE Contact Subject Where are you
these days? Content-Type application/sdp
Content-Length 124 v0 oajohnston 5462346
332134 IN IP4 sLet's Talk t0
0 cIN IP4 maudio 49170 RTP/AVP 0 3
  • SIP Responses
  • SIP Responses defined as (HTTP-style)
  • SIP-Version SP Status-Code SP Reason-Phrase CRLF
    (SPSpace, CRLFCarriage Return and Line Feed)
  • Example SIP/2.0 404 Not Found
  • First digit gives Class of response

  • SIP Responses Example
  • Required Headers
  • Via, From, To, Call-ID, and CSeq are copied
    exactly from Request.
  • To and From are NOT swapped!

SIP/2.0 200 OK Via SIP/2.0/UDP
host.wcom.com5060 From Alan Johnston
ltsipalan.johnston_at_wcom.comgt To Jean Luc Picard
ltsippicard_at_wcom.comgt Call-ID
  • SIP Responses Example
  • Typical SIP Response (containing SDP)

SIP/2.0 200 OK Via SIP/2.0/UDP From Alan Johnston
ltsipalan.johnston_at_wcom.comgt To Jean Luc Picard
ltsippicard_at_wcom.comgt Call-ID
m CSeq 1 INVITE Contact Subj
ect Where are you these days? Content-Type
application/sdp Content-Length
107 v0 opicard 124333 67895 IN IP4 sEngage! t0 0 cIN IP4 maudio 3456 RTP/AVP 0
  • Forking Proxy Example

SIP User Agent Client
SIP Proxy Server
SIP User Agent Server 2
SIP User Agent Server 1
100 Trying
404 Not Found
180 Ringing
180 Ringing
200 OK
200 OK
Media Stream
200 OK
  • SIP Headers - Partial List

  • SIP Headers - Continued

  • SIP Headers - Continued

  • 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

  • SIP Firewall Considerations
  • Firewall Problem
  • Can block SIP packets
  • Can change IP addresses of packets
  • TCP can be used instead of UDP
  • Record-Route can be used
  • ensures Firewall proxy stays in path
  • A Firewall proxy adds Record-Route header
  • Clients and Servers copy Record-Route and put in
    Route header for all messages

  • SIP Message Body
  • Message body can be any protocol
  • Most implementations
  • SDP - Session Description Protocol
  • RFC 2327 4/98 by Handley and Jacobson
  • http//
  • Used to specify info about a multi-media session.
  • SDP fields have a required order
  • For RTP - Real Time Protocol Sessions
  • RTP Audio/Video Profile (RTP/AVP) payload
    descriptions are often used

Session Description Protocol (SDP)
  • Convey sufficient information to enable
    participation in a multimedia session
  • SDP includes description of
  • Media to use (codec, sampling rate)
  • Media destination (IP address and port number)
  • Session name and purpose
  • Times the session is active
  • Contact information
  • Note indeed SDP is a data format rather than a

SDP Examples
  • v0
  • osisalem 28908044538 289080890 IN IP4
  • sSIP Tutorial
  • cIN IP4
  • t28908044900 28908045000
  • maudio 49170 RTP/AVP 0 98
  • artpmap98 L16/11025/2

  • SDP Examples (contd)

SDP Example v0 oajohnston 1-613-555-1212 IN
IP4 sLet's Talk t0 0 cIN IP4 maudio 49170 RTP/AVP 0 3
SDP Example v0 opicard 124333 67895 IN IP4 sEngage! t0 0 cIN IP4 maudio 3456 RTP/AVP 0
  • Another SDP Example (contd)

v0 oalan 1-613-1212 IN sSSE
University Seminar - SIP iAudio, Listen
only uhttp// ealan_at_wcom
.com p1-329-342-7360 cIN IP4 bCT128 t2876565 2876599 maudio
3456 RTP/AVP 0 3 atyperecvonly
  • Authentication Encryption
  • SIP supports a variety of approaches
  • end to end encryption
  • hop by hop encryption
  • Proxies can require authentication
  • Responds to INVITEs with 407 Proxy-Authentication
  • Client re-INVITEs with Proxy-Authorization
  • SIP Users can require authentication
  • Responds to INVITEs with 401 Unathorized
  • Client re-INVITEs with Authorization header

  • SIP Encryption Example

SIP/2.0/UDP host.wcom.com5060 From Alan
Johnston ltsipalan_at_wcom.comgt To Jean Luc Picard
ltsippicard_at_wcom.comgt Call-ID
m CSeq 1 INVITE Content-Length 224 Encryption
PGP version2.6.2, encodingascii q4aspdoCjh32a1_at_
  • PSTN Features with SIP
  • Features implemented by SIP Phone
  • Call answering 200 OK sent
  • Busy 483 Busy Here sent
  • Call rejection 603 Declined sent
  • Caller-ID present in From header
  • Hold a re-INVITE is issued with IP Addr
  • Selective Call Acceptance using From, Priority,
    and Subject headers
  • Camp On 181 Call Queued responses are monitored
    until 200 OK is sent by the called party
  • Call Waiting Receiving alerts during a call

  • PSTN Features with SIP
  • Features implemented by SIP Server
  • Call Forwarding server issues 301 Moved
    Permanently or 302 Moved Temporarily response
    with Contact info
  • Forward Dont Answer server issues 408 Request
    Timeout response
  • Voicemail server 302 Moved Temporarily response
    with Contact of Voicemail Server
  • Follow Me Service Use forking proxy to try
    multiple locations at the same time
  • Caller-ID blocking - Privacy Server encrypts
    From information

SIP User Location Example
SIP supports mobility across networks and devices
Qquality gives preference SIP/2.0 302 Moved
temporarily Contact
serviceIP,voice mail mediaaudio
duplexfull q0.7 Contact phone
1-972-555-1212 serviceISDN
mobilityfixed languageen,es, q0.5 Contact
phone 1-214-555-1212 servicepager
mobilitymobile duplexsend-only
mediatext q0.1 priorityurgent
descriptionFor emergency only Contact
Programming SIP
  • Examples
  • discard all calls from Monica during my
    business hours
  • redirect authenticated friends to my cell
    phone, anyone else to my secretary
  • if busy, return my homepage and redirect to
  • Users and third parties may program
  • SIP follows HTTP programming model
  • Mechanisms suggested in IETF CGI, Call
    Processing Language (CPL), Servlets

Call Processing Logic Example
SIP Mobility Support
Foreign Network
Home Network
1 INVITE 2 302 moved temporarily 3, 4
INVITE 5, 6 OK 7 Data
  • Global Wire and wireless
  • No tunneling required
  • No change to routing
  • For fast hand-offs use
  • Use Cellular IP or
  • Use DRCP

Dynamic Registration and Configuration Protocol
SIP Mobility
  • Pre-call mobility
  • MH can find SIP server via multicast REGISTER
  • MH acquires IP address via DHCP
  • MH updates home SIP server
  • Mid-call mobility
  • MH-gtCH New INVITE with Contact and updated SDP
  • Re-registers with home registrar

Need not bother home registrar Use multi-stage
registration Recovery from disconnects
Mobile IP Communications
  • Mobile IP Requirements
  • Transparency above L2
  • Move but keep IP address and all sessions alive
  • Mobility
  • Within subnet
  • Within domain
  • Global
  • AAA and NAIs
  • Location privacy
  • QoS for r.t. communications
  • Evolution of Wireless Mobility
  • Circuit Switched Mobility
  • based on central INs
  • LAN-MAN Cellular IP
  • Wide Area Mobile IP
  • Universal (any net) SIP

Presence, Instant Messaging and Voice
IP SIP Phones and Adaptors
  • Are Internet hosts
  • Choice of application
  • Choice of server
  • IP appliance
  • Implementations
  • 3Com (2)
  • Cisco
  • Columbia University
  • Mediatrix (1)
  • Nortel (3)
  • Pingtel

H.323/SIP Comparison
H.323 vs. SIP Basic Call Control
H.323 vs. SIP Advanced features
H.323 vs. SIP Scalability
H.323 vs. SIP Extensibility of functionality
H.323 vs. SIP Ease of customization
H.323 vs. SIP Ease of customization
H.323 vs. SIP Ease of Implementation
Summary SIP vs H.323
Internet Drafts
  • Session Timers in the Session Initiation Protocol
  • Caller Preferences for the Session Initiation
    Protocol (SIP)
  • Guidelines for Authors of Extensions to the
    Session Initiation Protocol (SIP)
  • The Stream Control Transmission Protocol as a
    Transport for for the Session Initiation Protocol
  • The Session Inititation Protocol (SIP) 'Replaces'
  • The SIP Referred-By Mechanism
  • Compressing the Session Initiation Protocol
  • Session Initiation Protocol Extension to Assure
    Congestion Safety
  • A Mechanism for Content Indirection in Session
    Initiation Protocol (SIP) Messages
  • The Session Inititation Protocol (SIP) 'Join'
  • SIP Authenticated Identity Body (AIB) Format
  • S/MIME AES Requirement for SIP
  • An Extension to the Session Initiation Protocol
    for Request History Information
  • Communications Resource Priority for the Session
    Initiation Protocol (SIP) Indicating User Agent
    Capabilities in the Session Initiation Protocol
  • Connection Reuse in the Session Initiation
    Protocol (SIP)
  • The Internet Assigned Number Authority Universal
    Resource Identifier Parameter Registry for the
    Session Initiation Protocol
  • The Internet Assigned Number Authority Header
    Field Parameter Registry for the Session
    Initiation Protocol
  • Session Initiation Protocol (SIP) Extension for
    Event State Publication
  • Interactions of Preconditions with Session
    Mobility in the Session Initiation Protocol (SIP)

  • The SIP INFO Method (RFC 2976)
  • MIME media types for ISUP and QSIG Objects (RFC
  • SIP Session Initiation Protocol (RFC 3261)
  • Reliability of Provisional Responses in SIP (RFC
  • SIP Locating SIP Servers (RFC 3263)
  • SIP-Specific Event Notification (RFC 3265)
  • DHCP Option for SIP Servers (RFC 3361)
  • Hypertext Transfer Protocol (HTTP) Digest
    Authentication Using Authentication and Key
    Agreement (AKA) (RFC 3310)
  • The Session Initiation Protocol UPDATE Method
    (RFC 3311)
  • Integration of Resource Management and SIP (RFC
  • Internet Media Type message/sipfrag (RFC 3420)
  • A Privacy Mechanism for the Session Initiation
    Protocol (SIP) (RFC 3323)
  • Private Extensions to the Session Initiation
    Protocol (SIP) for Asserted Identity within
    Trusted Networks (RFC 3325)
  • Session Initiation Protocol Extension for Instant
    Messaging (RFC 3428)
  • The Reason Header Field for the Session
    Initiation Protocol (SIP) (RFC 3326)Session
    Initiation Protocol Extension for Registering
    Non-Adjacent Contacts (RFC 3327)
  • Security Mechanism Agreement for the Session
    Initiation Protocol (SIP) Sessions (RFC 3329)
  • Private Session Initiation Protocol
    (SIP)Extensions for Media Authorization (RFC
    3313)The Session Initiation Protocol (SIP) Refer
    Method (RFC 3515)
  • Dynamic Host Configuration Protocol
    (DHCPv6)Options for Session Initiation Protocol
    (SIP) Servers (RFC 3319)
  • An Extension to the Session Initiation Protocol
    (SIP) for Symmetric Response Routing (RFC 3581)

Relevant IETF Working Groups
  • Audio/Video Transport (avt) - RTP
  • Differentiated Services (diffserv) QoS in
  • IP Telephony (iptel) CPL, GW location, TRIP
  • Integrated Services (intserv) end-to-end QoS
  • Media Gateway Control (megaco) IP telephony
  • Multiparty Multimedia Session Control (mmusic)
    SIP, SDP, conferencing
  • PSTN and Internet Internetworking (pint) mixt
  • Resource Reservation Setup Protocol (rsvp)
  • Service in the PSTN/IN Requesting InTernet
    Service (spirits)
  • Session Initiation Protocol (sip) signaling for
    call setup
  • Signaling Transport (sigtran) PSTN signaling
    over IP
  • Telephone Number Mapping (enum) surprises !
  • Instant Messaging and Presence Protocol (impp)

  • SIP Summary
  • SIP is
  • Relatively easy to implement
  • Gaining vendor and carrier acceptance
  • Very flexible in service creation
  • Extensible and scaleable
  • Appearing in products right now
  • SIP is not
  • Going to make PSTN interworking easy
  • Going to solve all IP Telephony issues (QoS)

  • Book on Internetworking Multimedia
    by Jon Crowcroft, Mark Handley, Ian Wakeman, UCL
    Press, 1999 by Morgan Kaufman (USA) and Taylor
    Francis (UK)
  • RFC 3261 SIP Session Initiation
  • http//
  • The IETF SIP Working Group home page
  • http//
  • SIP Home Page
  • http//
  • Papers on IP Telephony
  • http//