Janus WebRTC Server: SIP Gateway Demo (Sofia)

# Plugin Demo: SIP Gateway (Sofia)

### 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 Sofia-based SIP plugin: in case you're interested in the libre-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 demo, as the plugin it makes use of, has a few issues, right now. Specifically, it has only been tested with the widespread Asterisk PBX (extended with our Opus and VP8 support patch), even though it should work fine with others as well. Besides, video has not been tested yet and as such may not work as expected.

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

Use Video

### You

### Remote UA