[Freeswitch-users] Strange media behavior on WebRTC

Anthony Minessale anthony.minessale at gmail.com
Tue Jun 17 17:26:31 MSD 2014


For what it's worth, opus usually costs about the same bandwidth of ulaw.
If you are using pstn callers anyway ulaw is fine.
On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar at gmail.com> wrote:

> 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 at 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
>> brian at freeswitch.org
>>
>>
>> Twitter: @FreeSWITCH , @briankwest
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>>  T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
>> iNUM: +883 5100 1420 9001
>> ISN: 410*543
>> Skype:briankwest
>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>>
>>
>>
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>> On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar at 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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>
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>
> 
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