Support for legacy sip endpoints in videoroom plugin of Janus

[rammohan madhusudan]

Was experimenting with videoroom plugin that is there as part of Janus repo for webrtc based video conferencing.I also see that there is plugin developed for connecting to legacy SIP endpoints.

As i understand, if video room conference needs to created, we create a session in Core and subsequently videoroom plugin handle is attached to that session.

We want to experiment by adding a legacy SIP endpoint to this room.This would mean it would require transcoding as well.

Can the SIP plugin interact with Video room and join the conference?Does the janus video room plugin needs to enhanced to support this use case.

Is there a way this can be achieved.Any suggestions or pointers will help.

-Thanks,

Rammohan

[Diego Requena González]

I think here is explained what the plugin can and cannot do: https://github.com/meetecho/janus-gateway/blob/master/html/siptest.html#L51

Hope it helps.

Best regards.
D.

[Lorenzo Miniero]

Removed the other duplicate post.

The VideoRoom plugin doesn't use or understand SIP: it wouldn't make much sense anyway, considering all media is relayed in the VideoRoom, and very few SIP endpoints understand multiple media streams in a single session. If you want SIP and VideoRoom users to interact with each other, either go hybrid (e.g., VideoRoom for video and SIP for audio through a conference bridge like Asterisk or Freeswitch) or bridge the VideoRoom streams to/from SIP yourself (which is possible, as we did it for some customers, but harder as it requires ad-hoc development and media mixing/management).

L.