<div dir="ltr">If you're sure it's not an issue with the client implementation open an issue here:<div><a href="https://github.com/signalwire/freeswitch/issues">https://github.com/signalwire/freeswitch/issues</a> . Add pcap traces and FS logs taken with "amrwb_debug on" (fs_cli).<br></div><div>What happens if a 3rd member is added to the conference ? Does the 3rd member hear distorted audio too ? </div><div>If FS issue we need to figure out if the issue is on decoding or encoding. </div><div>When investigating be aware of the octet-align (OA) setting in SDP. It may be an offer/answer issue when we get an BE stream but we decode OA.</div><div> </div><div> </div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Sep 25, 2020 at 6:10 PM Rafał Telaczyński <<a href="mailto:rafal.telaczynski@jtendo.com">rafal.telaczynski@jtendo.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<p><font size="-1">Hi All,<br>
<br>
We are using FreeSWITCH to put 2 members on a conference and
record it. <br>
The problem is that when AMR-WB codec is used, the second member
to join the conference hears distorted sound(sometimes it lasts
couple seconds, in some cases it lasts until the end of the
call). There is no such sound heard by first member participant,
nor is it generated by it(verified by decoding audio from RTP
packets sent from first member participant towards FreeSWITCH).
Moreover, distorted sound to second member can be heard after
decoding RTP packets sent from FreeSWITCH towards second
conference participant - proof that it is not the issue with
second participant phone or so. <br>
Intrestingly, there is no such issue when AMR codec is
used(tested by removing AMR-WB from global_codec_prefs) - the
sound is clear for both participant throughout the whole
duration of the conference.<br>
<br>
Considering that everything is OK with AMR codec used, it seems
like the issue is not with conference itself, only using AMR-WB
during conference. We tested with some of the FreeSWITCH
parameters related to RTP(e.g. auto-rtp-bugs,
rtp-rewrite-timestamps, send_silence_when_idle, etc.) but it did
not help. <br>
Is it known issue? Did anyone faced something similar before?
What else can be done to make it work? Please, let me know if
you have any questions regarding the topic/need more info from
my side.<br>
<br>
Thanks, Rafal</font></p>
</div>
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