[Freeswitch-users] Strange media behavior on WebRTC

Wed Aug 6 00:51:32 MSD 2014

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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>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
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>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
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>>>>>>>>
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>>>>>>> 
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>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
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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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>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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:
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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
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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/  ☞
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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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> <http://freeswitch.org/g+>*
>
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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