We have noticed an odd issue which have kept us occupied most of today.
Since the Chromium rollout of M145 we have seen issues with one-way audio with our WebRTC clients. We receive RTP packages according to webrtc-internals but the AudioLevel for inbound-rtp remains constant at 0.
It appears to break in Asterisk 20.16.0 (from what we have tested), but works in instances across our farm running versions 20.12.0, 20.18.0 and 20.18.2.
We’re not asking for help, more like a shout-out to anyone else out there experiencing issues with audio since this release. Chromium changelogs do not really give much clue, and I cannot find any RTP related fix in the Asterisk changelog between 20.16.0 and 20.18.0 that would “fix” this.
We noticed this from user reports, and they all had in common that they just now updated their Chrome version to M145.
What do you all think?
Could it be that Chrome started caring about this? Our examples are not related to transfers though.