JsSIP
Template:Short description Template:DMCA
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| JsSIP | |
|---|---|
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| Initial release | 2011 |
| Written in | JavaScript |
| Engine | Page Template:Plainlist/styles.css has no content.Template:EditAtWikidata |
| Type | WebRTC |
| License | MIT |
| Website | jssip |
| Repository | github |
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JsSIP is a library for the programming language JavaScript. It takes advantage of SIP and WebRTC to provide a fully featured SIP endpoint in any website. JsSIP allows any website to get real-time communication features using audio and video. It makes it possible to build SIP user agents that send and receive audio and video calls as well as and text messages.[2]
General features
- SIP over WebSocket transport
- Audio-video calls, instant messaging and presence
- Pure JavaScript built from the ground up
- Easy to use and powerful user API
- Works with OverSIP, Kamailio, and Asterisk servers
- SIP standards
Standards
JsSIP implements the following SIP specifications:
- Template:IETF RFC — SIP: Session Initiation Protocol
- Template:IETF RFC — SIP Update Method
- Template:IETF RFC — The Reason Header Field for SIP
- Template:IETF RFC — SIP Extension Header Field for Registering Non-Adjacent Contacts (Path header)
- Template:IETF RFC — SIP Extension for Instant Messaging (MESSAGE method)
- Template:IETF RFC — Session Timers in SIP
- Template:IETF RFC — Managing Client-Initiated Connections in SIP (Outbound mechanism)
- Template:IETF RFC — Essential Correction for IPv6 ABNF and URI Comparison in RFC 3261
- Template:IETF RFC — Correct Transaction Handling for 2xx Responses to SIP INVITE Requests
- Template:IETF RFC — The WebSocket Protocol as a Transport for SIP
Interoperability
SIP proxies, servers
JsSIP uses the SIP over WebSocket transport for sending and receiving SIP requests and responses, and thus, it requires a SIP proxy/server with WebSocket support. Currently the following SIP servers have been tested and are using JsSIP as the basis for their WebRTC Gateway functionality:
- FreeSWITCH
- FRAFOS ABC WebRTC Gateway Script error: No such module "webarchive".
- OverSIP
- Kamailio
- Asterisk
- reSIProcate and repro
WebRTC web browsers
At the media plane (audio calls), JsSIP version 0.2.0 works with Chrome browser from version 24. At the signaling plane (SIP protocol), JsSIP runs in any WebSocket capable browser.
License
JsSIP is provided as open-source software under the MIT license.[3]
References
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- ^ Page Module:Citation/CS1/styles.css has no content."Releases". versatica/JsSIP. JsSIP. Retrieved 2 February 2017 – via GitHub.
- ^ Page Module:Citation/CS1/styles.css has no content."WebRTC:How and Why?" (PDF). FRAFOS. 12 January 2015. Archived from the original (PDF) on 12 June 2016. Retrieved 27 January 2015.
- ^ Page Module:Citation/CS1/styles.css has no content."JsSIP License".
External links
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