Janus WebRTC Server (multistream): NoSIP (SDP/RTP)

# Plugin Demo: NoSIP (SDP/RTP)

### Demo details

This is a demo that complements the one showcasing the SIP plugin. In fact, while the SIP plugin allows you to not worry about SIP details, which are implemented within the plugin itself, the NoSIP plugin doesn't mess with signalling itself, leaving it up to the application. As such, it provided an alternative to those that still want to interact with a legacy infrastructure (e.g., a pre-existing SIP-based one), but still want to be able to have control on the signalling themselves, rather than completely delegating it to the SIP plugin.

All this plugin does, as a consequence, is taking care of the translation between WebRTC empowered SDPs, and barebone SDPs that can be used with legacy peers. The barebone SDPs the plugin generates are crafted so that media is handled by the plugin itself, thus implementing the same RTP/RTCP gateway functionality the SIP plugin provides, but without the constraint of the signalling. It is up to the appplication to transport a generated offer in whatever signalling they want to use (e.g., SIP, IAX, XMPP, etc.) and make sure the offer/answer from the peer is passed to the plugin, so that the session can be completed.

Considering this plugin is very much generic and signalling-agnostic, this demo does NOT involve any signalling at all. On the contrary, it will show how a WebRTC peer can establish a session with another WebRTC peer (for the sake of simplicity located in the same page) by passing through the RTP/RTCP gatewaying functionality. This should as a result make it easier for you to understand how a NoSIP caller and a NoSIP callee would need to be implemented. The barebone SDPs generated/processed as a consequence will be displayed as a proof of concept.

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

### Caller

### Callee