Dragos Oancea
Dragos Oancea
Thank you for your answer and for pointing out the code. No seeking support needed for the scenarios I have in mind: playing opus files for IVRs (when the caller...
you will need to split it in two, since there are actually 2 different issues considered by this commit. so do one issue per commit. On the issue with the...
how often do you see it ? can you provide full log ?
did u try master branch ? this is probably fixed by: ``` commit 6c87ed491597fb5e30935d8309aa7e0c3aa9e18f Author: Michael Jerris Date: Fri Dec 17 13:55:11 2021 -0700 [Core] RFC 3264 says once we...
your statements seem inaccurate. without a network trace in pcap format we can't say anything. About the ACK problem: it can even be a firewall issue or SIP ALG -...
This is an English-only project, so the bug description should be in English. You're supposed to attach logs, not copy-paste on the GH issue. We'll need network trace in pcap...
and exit-sound is okay ? are the global variables recognized in exit-sound ? how about if you use channel variable "conference_moh_sound" ?
can you try this PR ? https://github.com/freeswitch/sofia-sip/pull/130 . you must rebuild and reinstall sofia-sip. then rebuild FS. the bug looks related to the code resolving ipv6 / AAAA
I pushed another commit and rebased the branch, so you'll get more fixes. please test.
this sort of core changes in RTP are hard to take without a proper unit-test, perhaps you can get inspired from https://github.com/signalwire/freeswitch/blob/master/tests/unit/switch_rtp_pcap.c and make an unit-test for this ? there...