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element-hq / element-web

A glossy Matrix collaboration client for the web.
https://element.io
GNU Affero General Public License v3.0
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Audio / video call debugging info #11511

Open jryans opened 4 years ago

jryans commented 4 years ago

(Extracted from https://github.com/vector-im/riot-web/issues/11368.)

Diagnosing STUN, TURN, WebRTC, media, etc. when something breaks can be quite challenging. To make this easier, we should add some kind of debugging UI focused on calls, which could display things like:

It would be even better if we made easy to select for copying into issues about calls as a way to gather up key facts (similar to Firefox's about:support for including system details).

ghost commented 4 years ago

@jryans I again (#11368) have VoIP connection problems for whose analysis logs would be very helpful. Would be great to at least have some raw logs.

tristanlins commented 4 years ago

I am also trying to set up my own Matrix Server + TURN Server and am failing with the voice and video calls. It seems to me that my TURN server is not being used at all, but I cannot understand why. Any kind of debugging option would be helpful.

ghost commented 4 years ago

Check if your TURN is running. If you are not sure if TURN is used, watch your syslog files on the server while establishing a connection. (grep "turnserver") Also you can check on one client in Firefox which IPs are offered for communication by the clients by opening "about:webrtc" in your browser's address bar, establish a call an refresh the tab to examine the "ICE Candidates" (Should be local IP, Public IP of Client (behind NAT), and IP of the Turn Server).

If the TURN server is not being used your yaml configuration file for Synapse may not be correct. Be aware that indentation is relevant. Be sure to use the same secret in both config files.

ghost commented 4 years ago

You may also want to list open ports on the server and/or scan if the relevant port is open for public.

tristanlins commented 4 years ago

My TURN server works. Test with turnutils_uclient was successful. I take a look what about:webrtc says.

grinapo commented 4 years ago

For video/audio problems it would be good to see whether it's a bandwidth, a packet loss or something else (browser, local camera, audio driver, …) problem.

Would be nice the possibility to enable

display on p2p audio/video calls.

A network problem indicator for the masses (a red worrying admin face worried ) may be also useful.

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