Janus WebRTC Server: SIP Gateway Demo (libre)

# Plugin Demo: SIP Gateway (libre)

### Demo details

This demo shows how you can make use of the SIP plugin to interact with a SIP Proxy (e.g., Kamailio) or PBX (e.g., Asterisk) in order to place or receive calls to and from other SIP clients. Specifically, it uses the libre-based SIP plugin: in case you're interested in the Sofia-based one, check this other demo instead. Notice that both plugins 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! This plugin is currently WIP, and so may not always work as expected. Considering the Sofia-based plugin has been around for much longer, and as of today has been used by more people in production, you may want to stick to that if what you're looking for is stability.

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

Use Video

### You

### Remote UA