I got a problem about VDO Call on WebRTC.

1. Brekeke Product Name and Version:
Brekeke SIP Server, Version 3.7.7.8, Advanced

2. Java version:
openjdk version "1.8.0_181"
OpenJDK Runtime Environment (build 1.8.0_181-b13)
OpenJDK 64-Bit Server VM (build 25.181-b13, mixed mode)

3. OS type and the version:
Red Hat Enterprise Linux Server release 6.6 (Santiago) 2.6.32-504.el6.x86_64

4. UA (phone), gateway or other hardware/software involved:
-
5. Your problem:
I got a problem about VDO Call on WebRTC, it's sometime well and sometime bad. And I'm not sure what's happen? In this case the RTP Relay was enabled.

I would like to know how to enable about RTP log or/and could you advise me whatever I need to do that help me find the root cause?

And If you have any advises please let me know.

Thank you
Yacht

Which WebRTC based SIP client are you using?

Generally WebRTC client can discover a way to exchange RTP packets automatically with ICE so you don't have to use "RTP-Relay".

Try the following settings at [Configuration]->[RTP] page.
[RTP relay] = "auto"
[RTP relay even with ICE] = "no"

If it doesn't work, try [RTP relay even with ICE] ="auto"