[jitsi-dev] Jigasi feature request - send RTP hole punching packets when SIP peer is connected


Using a crude POC I am creating conferences that only involve SIP client participants, however I am not seeing any conferenced audio media flowing apart from the SIP endpoint to SBC direction.
Our audio-only SIP endpoints are connected through an SBC that acts as a relay (no ICE involved).
The SBC is configured to not relay RTP in any direction until it receives RTP from both media legs of the call.
And it appears that Jigasi also will not forward RTP until the conference is receiving & producing media from at least one participant.
Therefore we are ending up in this no-audio situation (bit of an RTP Mexican stand-off!)

After some hacking I have managed to get media flowing for SIP-only conference calls.
The hack utilises CallPeerMediaHandler.sendHolePunchPacket() which will be called on behalf of the call when GatewaySession.CallPeerListener() sees that the peer SIP state is CallPeerState.CONNECTED.

So the feature request is to add this hole punching capability roughly as described above and make it configurable (enable/disable) and support the kinds of media that require its services (AUDIO or VIDEO).

Thx, ToM