Hello, We are testing webrtc via mod_verto with the verto_communicator demo. After some initial troubles with certificates, we have it running from Chrome and Firefox. Very nice! But, when we leave the connection on for some time, the audio in the direction from the browser to a phone becomes increasingly longer. We have clocked ~11 seconds, if we leave it on long enough. The audio in the other direction has a normal IP telephony delay. We have not found any browser dependencies. The FreeSWITCH version is 1.10.1 (latest stable release). The phone end has been a Blink softphone, or a trunk from a Cisco UCM. The latter negotiated mulaw PCM with a 20 ms packet size. We have tried different server OS:es: * Windows Server 2019 gives the fastest buildup of delay. * Debian installed in a Windows Substrate for Linux gives a slower buildup of delay, but it does continuously increase. * CentOS virtualized in VMWare doesn't have any buildup of delay, and the delay is a normal IP telephony delay. * So, the more Windows, the worse... I have a Wireshark trace, and I can get a TLS stream from it. I can install a key file to decode the stream, but the configuration wants a protocol, and I don't know what to enter there. There are no dissectors for Websocket. So I'm at a bit of a loss, as to how to debug this. I don't see anything obvious in the freeswitch.log. Questions: * Has anyone used webrtc on Windows? * Has anyone had the same problem? * What can be done to debug the problem? Turn on some logging? We could package a Linux machine to our customers, but as everything else is on Windows, it would be nice to make it work. Best regards, Jaan