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!.
• 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 3311 “SIP UPDATE Method”
• 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 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”