Totally agree about the desire to figure it out.  If we can do it, that would great.

PCMU vs OPUS is a separate issue from the noise after beep.

The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.  

However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well.  Easily reproducible on vanilla config on Windows 7.  Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.

Not sure I understand the remark about the ability to hear and understand everyone in the conference.  I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP).  Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case.  Will it?  

Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses?  I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it.  My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.



On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference?  Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it.  Its unacceptable to say 'Oh set it to 12000 it fixes it'.  The reason you have nobody complaining is they've never really experienced WebRTC properly.

You should also be using 1.0.1h, NOT g.

--
Brian West


Twitter: @FreeSWITCH , @briankwest














On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:

My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?

My system is in production already and nobody's been complaining about voice quality yet.



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