Anton Olofsson

I can confirm that recorded sessions still suffer from this issue on v1.10.11, and reverting https://github.com/signalwire/freeswitch/commit/47c5c8f3e8957c37fad5feeeb791375d05992b93 as suggested by @shaunjstokes resolves the issue.

Although the real question is why are there packages being generated at all? Inserting packages into the audio stream as soon as there are seemingly missing packets will negate the...

This fixes an issue where an unexpected payload is inserted into the wrong decoder after momentarily switching codecs (even though there was no need). This was made apparent after changes...

@emaktech Did you find any solution for this? We're suffering from what looks to be the same issue. Version 1.10.8

All of this is 100% just a matter of a missing marker bit when A's ringback stream is switched for the actual media stream from B - it's super apparent...

Further investigations indicate that Chrome (version 120) does not flush the jitter buffer and reset the stream upon receiving a marker bit, and thus a change of SSRC seems to...

@jakubkarolczyk Sorry for the delay, but here we go! I got one such log, at the moment when ringback starts being generated for the caller: `2024-02-27 14:02:20.610384 91.43% [INFO] switch_rtp.c:1565...

Interesting indeed! Can you share some lines of debug log context around your second hit @jakubkarolczyk?

@jakubkarolczyk `rtp_rewrite_timestamps` is not used

Hey there, I have a PR up which may be of interest to you. https://github.com/signalwire/freeswitch/pull/2275 I just came across your issue while browsing and it sounded very familiar. In my...