restcomm

Webrtc and restcomm

I'm new to restcomm, I did a small test with Restcomm Olympus and tried to connect Restcomm to a SIP client. It worked, but I do have many questions ref Restcomm webrtc support:

  1. Can I use Restcomm to make a call from SIP phone to webrtc browser (how will it work with Restcomm)
  2. Does Restcomm support Datachannels?
  3. Does it also work with browsers like Safari/Microsoft Edge?
  4. Does it support things like Trickle-ICE, can I integrate a TURN server to speed up call setup.
  5. Can Restcomm do some kind of transcoding between HD and non HD codecs?
  6. What capabilites Restcomm has for video support? I have tried to use Asterisk in the past for the video, but it isn't great.
Solution

    1. It will be available in the next version of Restcomm (Restcomm 7.6.0)

    2. Yes. You can use the WebRTC SDK that supports DataChannels https://github.com/RestComm/webrtcomm

    3. Not unless you use a plugin such as the Temasys plugin. It's a limitation of the browsers not implementing the WebRTC standard more than a Restcomm issue here

    4. Not yet, but we would welcome any contributions to that effort. You can use a TURN Server.

    5. Restcomm doesn't support HD Codecs at the moment and for the foreseeablee future as we may go with SFU type of solutions to better scale rather than using transcoding which would be very hard to scale for video anyway. We would welcome any contributions to that.

    6. You can do Peer to Peer Video calls but the Restcomm Media Server doesn't support video yet. It is planned on the roadmap with the intent of applying SFU as the solution (so no transcoding). We would welcome any contributions to that effort as well.