[Freeswitch-users] One way Jitter Buffer

[Anthony Minessale]

Enable jitter buffer at 20 ms on both legs and call it a day, its adaptive.

On Wednesday, June 3, 2015, Kamil Nigmatullin <kamil.nigmatullin at gmail.com>
wrote:

> What I mean, I have this scheme:  [Android Phone on 3G]  <--->
> [Freeswitch]  <---->[DID Provider]. When Android user calls DID Provider, I
> want to apply Jitter Buffer on Freeswitch so that voice (stream) coming
> from Android become smoother. On the same time, I want that the stream from
> DID provider of the same call was not affected by Jitter Buffer and will be
> delivered to Android Phone unchanged.
>
> Thanks
>
> 2015-06-03 21:09 GMT+06:00 Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>>:
>
>> you don't buffer outgoing rtp.  jitter is introduced on the wire after it
>> leaves the box so buffering outgoing streams would only delay audio and not
>> add any value.
>>
>>
>> On Wednesday, June 3, 2015, Kamil Nigmatullin <
>> kamil.nigmatullin at gmail.com
>> <javascript:_e(%7B%7D,'cvml','kamil.nigmatullin at gmail.com');>> wrote:
>>
>>> Dear All,
>>>
>>> We have developed our own sip application that works with Freeswitch.
>>> So, on incoming rtp stream we can ajust jitter buffer on client side, but
>>> on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a
>>> way to switch Jitter Buffer only for outgoing stream in Freeswitch
>>>
>>> Thanks
>>>
>>> --
>>> Kamil Nigmatullin
>>> Tel: 77272323748
>>> mob: 7 (707) 2517003
>>> Skype: kamil.nigmatullin
>>>
>>
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>
>
>
> --
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003
> Skype: kamil.nigmatullin
>


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

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