I got a problem about VDO Call on WebRTC.

[punthape]

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

[Mohney]

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"