[Freeswitch-users] Strange media behavior on WebRTC

Tue Aug 5 21:48:54 MSD 2014

Thanks Tony!

Will there be any side effects if I use this setting?

Oleg
On Aug 5, 2014 12:06 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> Looks like if you add
>
> <action application="set" data=
> "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>
> before you call answer or the conference it fixes it.
>
> So the media engine is getting confused by our timestamps at the beginning
> of the call.
>
>
>
>
>
>
> On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Thanks Michael!  I've been testing 12000 and 16000 for a couple of days
>> now.  So far 12000 seems to use marginally (very marginally) less CPU than
>> 16000.  I'll monitor CPU for several more days before settling on one or
>> the other.
>>
>>
>> On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> if you have to up the rate to get around this issue, setting it to 16k
>>> instead of 12k when your endpoints are all 8k should use less cpu.
>>>
>>> On Jun 17, 2014, at 1:26 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>> 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
>>>>>
>>>>> <whmcslogo.png>
>>>>>
>>>>> 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
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>>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 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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>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
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