Hello everyone. There’s this unexplainable problem with WebRTC: right after the call starts, the voice sounds garbled, choppy, and very hard to understand for about 5–10 seconds, then everything is fine. Occasionally, there might be another 5 seconds of artifacts. However, jitter is normal (15–30 ms), there’s no packet loss. And this doesn’t happen to everyone — the settings are the same, but some users experience artifacts from time to time, while for others everything works fine. Could you please advise what this might be? Maybe there’s some specific setting for such cases?
This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.