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SIP Protocols

  • SIP Protocols

http://www.packetizer.com/voip/sip/standards.html

Core SIP Documents
RFC 2543SIP: Session Initiation Protocol (obsolete)
RFC 3261SIP: Session Initiation Protocol

SDP-Related Documents
RFC 2327Session Description Protocol (SDP)
RFC 3264An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 3266Support of IPv6 in SDP
RFC 3388Grouping Media Lines in SDP
RFC 3407Session Description Protocol (SDP) Simple Capability Declaration
RFC 3556SDP Bandwidth Modifiers for RTCP Bandwidth
RFC 3605Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)
RFC 3890A Transport Independent Bandwidth Modifier
RFC 4091An Alternative NAT Semantics for SDP
RFC 4145TCP-Based Media Transport in the SDP

RTP-Related Documents
RFC 3550RTP: Transport Protocol for Real-Time Applications
RFC 3551RTP Profile for A/V Conferences with Minimal Control
RFC 2198RTP Payload for Redundant Audio Data
RFC 2733An RTP Payload Format for Generic Forward Error Correction
RFC 2793RTP Payload for Text Conversation
RFC 2833RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2959Real-Time Transport Protocol Management Information Base
RFC 3389RTP Payload for Comfort Noise
RFC 3611RTP Control Protocol Extended Reports (RTCP XR)
RFC 3711The Secure Real-time Transport Protocol (SRTP)

HTTP-Related Documents
RFC 2616Hypertext Transfer Protocol -- HTTP/1.1
RFC 2617HTTP Authentication: Basic and Digest Access Authentication
RFC 3310HTTP Digest Authentication Using Authentication and Key Agreement (AKA)
RFC 3310HTTP Digest Authentication Using Authentication and Key Agreement (AKA)

MIME-Related Documents
RFC 1847Security Multiparts for MIME: Multipart/Signed and Multipart/Encrypted
RFC 2045MIME Part One: Format of Internet Message Bodies
RFC 2046MIME Part Two: Media Types
RFC 2047MIME Part Three: Message Header Extensions for Non-ASCII Text
RFC 2048MIME Part Four: Registration Procedures
RFC 2633S/MIME Version 3 Message Specification
RFC 3204MIME media types for ISUP and QSIG Objects
RFC 3420Internet Media Type message/sipfrag
RFC 3555MIME Type Registration of RTP Payload Formats

SIP Extension and Options
RFC 2976The SIP INFO Method
RFC 2848Extensions for IP Access to Telephone Call Services
RFC 3050CGI for SIP
RFC 3262Reliability of Provisional Responses
RFC 3263Locating SIP Servers
RFC 3265SIP-Specific Event Notification
RFC 3311UPDATE Method
RFC 3312Integration of Resource Management and SIP
RFC 3313Private SIP Extensions for Media Authorization
RFC 3319DHCPv6 Options for SIP Servers
RFC 3323A Privacy Mechanism for SIP
RFC 3324Short Term Requirements for Network Asserted Identity
RFC 3325Private Extensions to SIP for Asserted Identity within Trusted Networks
RFC 3326The Reason Header Field
RFC 3327Extension Header Field for Registering Non-Adjacent Contacts
RFC 3329Security Mechanism Agreement
RFC 3361DHCP-for-IPv4 Option for SIP Servers
RFC 3372SIP for Telephones (SIP-T): Context and Architectures
RFC 3398ISUP to SIP Mapping
RFC 3428SIP Extension for Instant Messaging
RFC 3455Private Header Extensions for 3GPP
RFC 3515The Session Initiation Protocol (SIP) Refer Method
RFC 3578Mapping ISUP Overlapped Signalling to SIP
RFC 3581Extension to SIP for Symmetric Response Routing
RFC 3608Extension Header Field for Service Route Discovery During Registration
RFC 3680SIP Event Package for Registrations
RFC 3840Indicating User Agent Capabilities in SIP
RFC 3841Caller Preferences for SIP
RFC 3842Message Summary and Message Waiting Indication Event Package
RFC 3856Presence Event Package
RFC 3857A Watcher Information Event Template-Package
RFC 3891"Replaces" Header
RFC 3903Event State Publication
RFC 3959Early Session Disposition Type
RFC 3960Early Media and Ringing Tone Generation
RFC 4028Session Timers in the Session Initiation Protocol (SIP)
RFC 4235An INVITE-Initiated Dialog Event Package for SIP
RFC 4244Extension for Request History Information
RFC 4320Actions Addressing Identified Issues with the SIP Non-INVITE Transaction
RFC 4411Extending the SIP Reason Header for Preemption Events
RFC 4412Communications Resource Priority for SIP
RFC 4488Suppression of SIP REFER Method Implicit Subscription

SIP Informational RFCs and BCP Documents
RFC 3087Control of Service Context using SIP Request-URI
RFC 3351User Requirements for SIP in Support of Speech/Hearing Impaired
RFC 3603Private SIP Proxy-to-Proxy Extensions for PacketCable Distributed Call Signaling
RFC 3702Authentication, Authorization, and Accounting Requirements for SIP
RFC 3824Using E.164 numbers with SIP
RFC 3911The SIP "Join" Header
RFC 3968IANA Registry for SIP Header Field
RFC 3969IANA Registry for SIP URI
RFC 3976Interworking SIP and IN Applications
RFC 4117Transcoding Services Invocation using 3PCC
RFC 4123SIP-H.323 Interworking Requirements
RFC 4168SCTP as a transport for SIP
RFC 4189Requirements for End-to-Middle Security for SIP
RFC 4240Basic Network Media Services with SIP
RFC 4245High-Level Requirements for Tightly Coupled SIP Conferencing
RFC 4317SDP Offer/Answer Examples
RFC 4321Problems Identified Associated with the SIP Non-INVITE Transaction
RFC 4353A Framework for Conferencing with SIP
RFC 4354SIP Event Package and Data Format for Push-to-Talk over Cellular (PoC) Service
RFC 4453Requirements for Consent-Based Communications in the SIP
RFC 4457SIP P-User-Database Private-Header (P-Header)
RFC 4458SIP URIs for Applications such as Voicemail and Interactive Voice Response (IVR)
RFC 4475SIP Torture Test Messages

SIP-Related Documents
RFC 3219Telephony Routing over IP (TRIP) (tutorial)
RFC 3320Signalling Compression
RFC 3321Signalling Compression - Extended Operations (informational)
RFC 3322Signalling Compression - Requirements and Assumptions (informational)
RFC 3486Compressing the Session Initiation Protocol (SIP)
RFC 3485SIP and SDP Static Dictionary for Signaling Compression
RFC 3725Best Current Practices for 3PCC in SIP
RFC 3764enumservice registration for SIP Addresses-of-Record
RFC 4077A Negative Acknowledgement Mechanism for Signaling Compression
RFC 40833GPP Release 5 Requirements on SIP
RFC 4092Using SDP Alternative NAT Semantics in SIP
RFC 4497Interworking between the SIP and QSIG

Directory Services Documents
H.350Directory Services Architecture for Multimedia Conferencing
H.350.4Directory Services Architecture for SIP


This page is maintained by Paul E. Jones. Please feel free to report an errors directly to me.


  • SIP Applicatoins

http://www.ekiga.org/

Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting.

  • Open Source SIP Libs


NameDescribe 
AsteriskAsterisk is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. 
MinisipMinisip is a SIP User Agent ("Internet telephone").
It can be used to make phone calls, instant message and videocalls to your buddies connected to the same SIP network.		 
oSIP stack (GNU)oSIP implements the Session Initiation Protocol (published by IETF as RFC 3261). It can provide signalling capabilities for multimedia applications (IP phones, etc.). It provides a fully usable parser for the SIP syntax and implements the "transaction layer" as defined in the draft. It also provides an SDP parser and extra features for the User Agent. It can be used to build both proxy and IP phones. 
*pjsip
*oldsite		Small footprint SIP stack  
PartysipSIP proxy server 
SIPfoundry.org  
SIP Express Router (SER) (iptel.org)SIP Express Router (SER) (iptel.org)
ptel.org SIP Express Router" (SER) is a high performance configurable, free server implementing Session Initiation Protocol (SIP).

It can act as registrar, proxy server, redirect server and server many roles in SIP networks, including PSTN gateway guard, SMS/Jabber gateway and application server.

http://www.iptel.org/ser/

iptel.org
Description: iptel.org collects related IETF documents and links to other resources related to SIP. The site offers free SIP accounts. SIP Express Router, free SIP server, is available from iptel.org as well.

 
sipsak

sipsak is a small command line tool for developers and administrators of Session Initiation Protocol (SIP) applications. It can be used for some simple tests on SIP applications and devices. sipsak is a “swiss army knife” for SIP developers.

sipsak is an open source tool available at http://sipp.sourceforge.net/. The home page for the tool is at http://www.sipsak.org/.

 
SUNSun also has a number of open source SIP projects:
  1. A Java SIP stack compliant to JSR32 (http://jain-sip.dev.java.net)
  2. A proxy server application and a presence/IM user agent that runs on JSR32 (http://jain-sip-presence-proxy.dev.java.net)
  3. A soft phone with IM capabilities that runs on JSR32 (http://sip-communicator.dev.java.net)
 
Vovidaa communications community site dedicated to providing a forum for open source software used in datacom and telecom environments 
Yate - Yet Another Telephony Engine

ATE is a telephony engine designed to implement PBX and IVR solutions for small to large scale projects.

Yate can be used as a:

  • VoIP server
  • VoIP client
  • VoIP to PSTN gateway
  • PC2Phone and Phone2PC gateway
  • H.323 gatekeeper
  • H.323 multiple endpoint server
  • H.323<->SIP Proxy
  • SIP session border controller
  • SIP router
  • SIP registration server
  • IAX server and/or client
  • IP Telephony server and/or client
  • Call center server
  • IVR engine
  • Prepaid and/or postpaid cards system
 
sofia-sipSofia-SIP - a RFC3261 compliant SIP User-Agent library. 

  • SIP Tools

Distributed SIP Analyzer

  • Site

http://www.sipforum.org/

http://www.sipfoundry.org/

  SIPfoundry, the leading Open Source Community dedicated to SIP Solutions and Development
Key efforts at SIPfoundry include:
 
The SIPfoundry Wiki provides additional documentation.

http://www.sipcenter.com/

Vovida http://www.vovida.org/

Welcome to Vovida.org - a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments. While we have predominantly included software to date that we have created, we are looking forward to adding more software from all of you as it is submitted to us for inclusion. If you are working on any type of software for broadband, wideband, or narrowband environments for cable, xDSL, wireless,

  • Documents


SIP: Creating next-generation telecom applications 
Application servers & Media Servers 
SIP products 
Learning Guide: SIP学习sip，有很多相关链接