Janus WebRTC Server (multistream): SIP Gateway Demo

# Plugin Demo: SIP Gateway

Want to learn more about the SIP plugin? Check the Documentation.

### Demo details

This demo shows how you can make use of the SIP plugin to interact with a SIP Proxy (e.g., Kamailio or OpenSIPS) or PBX (e.g., Asterisk or FreeSwitch) in order to place or receive calls to and from other SIP clients. Specifically, it uses the Sofia-based SIP plugin. Notice the plugin only exchange SIP messages from within the plugin itself: no SIP is done in JavaScript, except for references to SIP URIs.

When started, the demo will allow you to insert a minimum set of information required to REGISTER the web page as a SIP client at a SIP Proxy or PBX you specify. This will allow you to call SIP URIs, or receive calls through the SIP Server itself. During a call, you'll also be able to interact with the PBX via DTMF tones, e.g., to drive an Interactive Voice Response (IVR) menu that you're being presented with.

Note well! Please notice that, while audio support has been tested extensively, video hasn't as much, and as such may not work as expected.

Press the `Start` button above to launch the demo.

Use Video
You
Remote UA