[Freeswitch-users] Strange media behavior on WebRTC

Tue Aug 5 22:24:50 MSD 2014

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

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