SIP Introduction
|
SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to (and removed from) an existing session. SIP transparently supports name mapping and redirection services, which supports personal mobility - users can maintain a single externally visible identifier regardless of their network location. SIP should be used in conjunction with other protocols in order to provide complete services to the users. These architectures will include protocols such as the Real-time Transport Protocol (RTP) for transporting real-time data and providing QoS feedback, the Real-Time streaming protocol (RTSP) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) for describing multimedia sessions. |
SIP Protocol
- Overview of Operation
- Definitions
- SIP Messages
- General User Agent Behavior
- Registrations
- Initiating a Session
- Proxy Behavior
- Header Fields
- Response Codes
- SIP Session Setup Example - Cheat Sheet (download)
- SIP Registration Examples - Cheat Sheet (download)
SIP Tutorials
- SIP Cookbook - sip.edu
- SIP Technical Overview - Avaya
- SIP Protocol Overview - Radvision
- SIP Tutorial - TEKELEC
- Programming SIP - SIP Center
- SIP - Lecture slides of Henning Schulzrinne
- SIP - Network Sorcery
- NAT Traversal for Multimedia over IP Services - Newport Networks
- SIP, Security and Session Controllers - Newport Networks
- Lawful Intercept Overview - Newport Networks
- Secure Voice-over-IP by Markus and Christoph
- TR/87 Using CSTA for SIP phone UAs - ECMA
Standards
Detailed list as per Rosenburg's Hitchhikers guide to SIP
Portals
- The SIP Center
- SIP Technical Portal - Tech Invite
- pulver.com - "Voice" of IP Communications
- TMCnet - VoIP News and Internet Telephony
- VOIP Wiki - www.voip-info.org
- TELEPHONY Magazine
- VoIP Magazine Online
Implementations
- Open Source
- Asterisk | The Open Source PBX
- FreeSWITCH
- Kamailio SIP Server
- OpenSIPS Project
- SIP server for Windows
- Brekeke SIP Server
- SIP Express Router
- OpenXCAP - Free XCAP server
- OpenSBC
- Sippy B2BUA
- SIPfoundry
- The GNU oSIP library
- The eXtended osip library
- pjsip.org
- PartySIP
- SIP Express Router
- Java SIP
- libSRTP - SRTP library
- Commercial Servers
- Avaya
- Cisco
- Broadsoft
- Microsoft OCS
- Commercial Source Code
- Aricent
- Radvision
- M5T
- Data Connection
- Trillium
Soft Phones
- Xlite (xten)
- Kapanga Soft Phone
- Zoiper Softphone
- Windows Messenger - Microsoft
- SIP Multimedia PC Client - Nortel
- OpenWengo
- Express Talk
- Sippax
- Linphone (Linux)
- Twinkle (Linux)
Cisco
- SIP - Technology Support
- Quality of Service for VoIP
- Cisco IOS Voice Configuration Library
- Cisco Unified SIP SRST 4.0
Testing
Mailing Lists
Products
Links
- SIP Resources - Henning Schulzrinne
- SIP Resources - Jonathan Rosenberg
- SIP.edu
- Packetizer.com SIP
- SIP - Wikipedia
- VOIP Wiki - www.voip-info.org
Books
- 1. SIP by Alan B. Johnston
- 2. SIP Demystified by Gonzalo Camarillo
- 3. Internet Communications Using SIP by Henry Sinnreich, Alan B. Johnston
- 4. SIP Beyond VoIP by Henry Sinnreich, Alan B. Johnston
Maintained by Anil Edathara
Last Modified: February 21, 2010
| SIP Methods | RFC | |
| ACK | Acknowledge final response to Invite | 3261 |
| BYE | Terminate a session | 3261 |
| CANCEL | Cancel a previous request | 3261 |
| INFO | Mid-session signaling | 2976 |
| INVITE | Initiate a session | 3261 |
| MESSAGE | Allows the transfer of IMs | 3428 |
| NOTIFY | Event notification | 3265 |
| OPTIONS | Query to find the capabilities | 3261 |
| PRACK | Acknowledgement for Provisional responses | 3262 |
| PUBLISH | Publish event state | 3903 |
| REFER | Transfer user to a 3rd party | 3515 |
| REGISTER | Register with a SIP network | 3261 |
| SUBSCRIBE | Request asynchronous event notification | 3265 |
| UPDATE | Update parameters of a session | 3311 |
| Response Codes | ||
Provisional 1xx 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress Successful 2xx 200 OK 202 Accepted Redirection 3xx 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service Request Failure 4xx 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidden 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 408 Request Timeout 410 Gone 412 Conditional Request Failed 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 417 Unknown Resource-Priority 420 Bad Extension 421 Extension Required 422 Session Interval Too Small 423 Interval Too Brief 428 Use Identity Header 429 Provide Referrer Identity 436 Bad Identity-Info 437 Unsupported Certificate 438 Invalid Identity Header 480 Temporarily Unavailable 481 Call/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 489 Bad Event 491 Request Pending 493 Undecipherable 494 Security Agreement Required Server Failure 5xx 500 Server Internal Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Server Time-out 505 Version Not Supported 513 Message Too Large 580 Precondition Failure Global Failures 6xx 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable |
||