• SIP Protocols

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

Core SIP Documents

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

SDP-Related Documents

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

RTP-Related Documents

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

HTTP-Related Documents

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

MIME-Related Documents

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

SIP Extension and Options

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

SIP Informational RFCs and BCP Documents

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

SIP-Related Documents

RFC 3219	Telephony Routing over IP (TRIP) (tutorial)
RFC 3320	Signalling Compression
RFC 3321	Signalling Compression - Extended Operations (informational)
RFC 3322	Signalling Compression - Requirements and Assumptions (informational)
RFC 3486	Compressing the Session Initiation Protocol (SIP)
RFC 3485	SIP and SDP Static Dictionary for Signaling Compression
RFC 3725	Best Current Practices for 3PCC in SIP
RFC 3764	enumservice registration for SIP Addresses-of-Record
RFC 4077	A Negative Acknowledgement Mechanism for Signaling Compression
RFC 4083	3GPP Release 5 Requirements on SIP
RFC 4092	Using SDP Alternative NAT Semantics in SIP
RFC 4497	Interworking between the SIP and QSIG

Directory Services Documents

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

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

---

• SIP Applicatoins

• ekiga 

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

Name	Describe
Asterisk	Asterisk 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.
Minisip	Minisip 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
Partysip	SIP 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/.
SUN	Sun also has a number of open source SIP projects: A Java SIP stack compliant to JSR32 (http://jain-sip.dev.java.net) A proxy server application and a presence/IM user agent that runs on JSR32 (http://jain-sip-presence-proxy.dev.java.net) A soft phone with IM capabilities that runs on JSR32 (http://sip-communicator.dev.java.net)
Vovida	a 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-sip	Sofia-SIP - a RFC3261 compliant SIP User-Agent library.

---

• SIP Tools

Distributed SIP Analyzer

---

• Links

  • Klaus Darilion's bookmarks of VoIP !!!

  • columbia sip page
    

• 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:
 

• sipX, The open source SIP PBX for Linux
• repro, the SIP (Proxy) Server
• reSIProcate, a SIP reference stack implementation
• sipXphone, a SIP softphone for Windows and Linux environments
• sipXezPhone and sipXtapi, a comprehensive SIP user agent SDK
• SFTF, the SIP Forum Test Framework

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,

Applications
Apps Home
VOCAL
B2BUA
Load Balancer
MPEG4IP
Open G.729(A)
Open AMR
OpenOSP
RGL
SIPRG
SIPSet
SIPTiger
STUN Server
WinRTP

Protocols
Protocols Home
SIP
COPS
GDOI
GSML
MGCP
RTSP
RTP
RADIUS
SRTP

TRIP

• ---

• 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