Hello,

For those of you integrating with WebRTC and using a websocket control
plane like SIP.js how do you handle trickle ICE? If you use trickle
ICE any recommended RTP media servers (FreeSWITCH, Kurento, ....)?

SIP.js does some trick like delay XXX ms before sending out the offer
assuming the browser has collected all the ICE candidates, or maybe
even wait for the IceGatheringDone event: what do you think of such a
pattern?

I see that rtpengine source code mentions "trickle" in a few places,
but the rtpengine module doesn't make it clear how to send a ICE
candidate for an existing call.


Cheers
Anthony Alba

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