[Freeswitch-users] Strange media behavior on WebRTC

Wed Aug 6 01:43:29 MSD 2014

It was not happening on your webrtc demo when you originally tried to
reproduce it with 8765 extension.  Then it was not happening when I tried
it on Debian 7.  It's probably not worth investigating - I was just curious.

Thank you
Oleg
I have no idea really.  Maybe it did and you missed a step reproducing it?

On Tue, Aug 5, 2014 at 3:51 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Out of curiosity - why was this not happening on Debian 7?
>
> Thank you
> Oleg
> On Aug 5, 2014 3:29 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
> wrote:
>
>> 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
>>>>>>>>>> iNUM: +883 5100 1420 9001
>>>>>>>>>> ISN: 410*543
>>>>>>>>>> Skype:briankwest
>>>>>>>>>> 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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>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
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>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
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>>>>>>>>>
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>>>>>>>>
>>>>>>>> 
>>>>>>>> 
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>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
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>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
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>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>>>>>
>>>>>> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
>>>>>> http://twitter.com/FreeSWITCH
>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
>>>>>> <http://freeswitch.org/g+>*
>>>>>>
>>>>>> ClueCon Weekly Development Call
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>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
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>>>>>>
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>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
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>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>>>
>>>> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
>>>> http://twitter.com/FreeSWITCH
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>>>> <http://freeswitch.org/g+>*
>>>>
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>>>>
>>>>
>>>> _________________________________________________________________________
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>>>>
>>>> 
>>>> 
>>>>
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>>> _________________________________________________________________________
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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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>>
>> 
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>>
> _________________________________________________________________________
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>
> 
> 
>
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>

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

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