[Freeswitch-users] Strange media behavior on WebRTC

Anthony Minessale anthony.minessale at gmail.com
Wed Aug 6 00:25:12 MSD 2014 

No,  RTP timeout will still work.  I enabled this flag by default now when
webrtc is enabled since every browser seems to suffer from it.



On Tue, Aug 5, 2014 at 2:06 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> 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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>>>>>>>> Skype:briankwest
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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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>>>>>> 
>>>>>> 
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>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
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>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>
>>>>> 
>>>>> 
>>>>>
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>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>>>
>>>> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
>>>> http://twitter.com/FreeSWITCH
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>>>>
>>>>
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>>>>
>>>> 
>>>> 
>>>>
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>>
>>
>> --
>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>
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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
> _________________________________________________________________________
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>
> 
> 
>
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>


-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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