[Freeswitch-users] Strange media behavior on WebRTC

Tue Aug 5 23:06:04 MSD 2014

It won't disable rtp timeout though, right?

Thank you
Oleg
On Aug 5, 2014 1:34 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> No, it just disables the proper behavior of our rtp stack which is to have
> time still pass when not sending RTP.
> With the param enabled, the timestamps only go up when we are actually
> writing packets which is incorrect but does fix your problem ;)
>
>
>
> On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> 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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>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 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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>>>>>
>>>>
>>>>
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>>>
>>>
>>> --
>>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>>
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>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
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