We are using FOP2 with built-in WebRTC phone.
For some time we’ve been observing an issue with stopping RTP stream originated from the WebRTC endpoint. We traced the RTP traffic back and confirmed that the RTP packets just stop to leave the PC - it looks like that:
The 192.168.16.58 is the address of the local machine with WebRTC client that the dump was made on.
The 172.31.30.245 is our FreePBX 16 system running FOP2 2.31.40 (upgraded recently to check if issue persists).
From that point the issue looks like to be on the software level (somewhere between the WebRTC subsystem and audio devices) but I haven’t found any way to deeper look into it.
Chrome’s tools (chrome://webrtc-internals) give lots of information about the various aspects of the call but there’s very little on hardware media devices.
FOP2 WebRTC is also rather limited since there’s no way to inspect audio devices that are available and currently in use.
Agents commonly use USB headphones and we checked that plugging out and replugging in the headphones during ongoing call gives the same pcap output - the RTP originating from agent vanishes, but I’d like to have some more tools to look deeper in the media devices to trace the issue since USB audio devices are rather well implemented hardware and I wouldn’t expect issues like this from the hardware side.
So far I haven’t found any way to enable more debugging on the client level (other than mentioned chrome://webrtc-internals - which does not give me any hardware related information).
So, maybe anyone with more experience in the field could point me to a right direction?