Overview
JsSIP is a simple to use JavaScript library which leverages latest developments in SIP and WebRTC to provide a fully featured SIP endpoint in any website.
With JsSIP any website can get Real Time Communications features using audio, video and more with just a few lines of code.
Features
- SIP over WebSocket transport.
- Audio/video calls, instant messaging and presence.
- Lightweight! (~140KB).
- 100% pure JavaScript built from the ground up.
- Easy to use and powerful user API.
- Works with OverSIP, Kamailio and Asterisk servers.
SIP Standards
JsSIP implements the following SIP specifications:
- RFC 3261 “SIP: Session Initiation Protocol”
- RFC 3326 “The Reason Header Field for SIP”
- RFC 3327 “SIP Extension Header Field for Registering Non-Adjacent Contacts” (
Path
header) - RFC 3428 “SIP Extension for Instant Messaging” (
MESSAGE
method) - RFC 5626 “Managing Client-Initiated Connections in SIP” (
Outbound
mechanism) - RFC 5954 “Essential Correction for IPv6 ABNF and URI Comparison in RFC 3261”
- RFC 6026 “Correct Transaction Handling for 2xx Responses to SIP INVITE Requests”
- draft-ietf-sipcore-sip-websocket “The WebSocket Protocol as a Transport for SIP”