Closed greenfieldtech-nirs closed 6 years ago
You applied the patch for Native PLC. That patch is experimental. Please, undo that patch and try again.
A) If that was the cause, please continue there… B) If not, I need much more details to reproduce your issue:
Hmmmm... I don't recall adding that patch in, but could be. try and report back.
lost frame(s)
This log output came with the patch. Somehow in your scenario, the sequence numbers of the RTP packets are not +1 as expected. This should be visible via Wireshark as well. If this is the cause for your no-audio symptom, please, double-check what causes that. Are you using SIP Session Timers? Asterisk active them on default. However, many implementations are terrible wrong and reset all kind of stuff on the re-INVITE message.
We noticed the issue, seems like one of the experimental patches got in there and messed things up. Thanks for the assist.
Upon using Opus with Asterisk 14.7.X or Asterisk 15.5.0 an issue is exhibited, where the codec will simply stop transcoding in a certain direction. For example:
UA (A) ----- Opus -----> Asterisk ---- ulaw -----> PSTN Phone (B)
In the above case, after an unknown period of time the audio path from (A) to (B) is no longer audioable, while the path from (B) to (A) is still available.
A debug of Asteris and 'rtp set debug on' produced the following output:
Now, after that happens, we shortly see the following:
Which turns into:
Which at that point loses audio completely from (A) towards (B).
The issue is re-produceable and happens constantly, however, will happen at different intervals.