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.


  • SIP over WebSocket transport.
  • Audio/video calls, instant messaging and presence.
  • Lightweight!.
  • 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 3261SIP: Session Initiation Protocol”
  • RFC 3311SIP UPDATE Method”
  • RFC 3326 “The Reason Header Field for SIP
  • RFC 3327SIP Extension Header Field for Registering Non-Adjacent Contacts” (Path header)
  • RFC 3428SIP Extension for Instant Messaging” (MESSAGE method)
  • RFC 3515 “The SIP Refer Method”
  • RFC 3891 “The SIP Replaces Header”
  • RFC 4028 “Session Timers in SIP
  • RFC 5589 “The SIP Call Control – Transfer”
  • 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”
  • RFC 7118 “The WebSocket Protocol as a Transport for SIP