[Freeswitch-users] transcoding GSM with a high ptime

[Anthony Minessale]

The only negative side effect is that with no timer, the channel only
times off the inbound stream so if there is no rtp it would not
playback sound.  This is similar to how asterisk behaves by default.
It's not noticeable at all during bridge but could be noticeable while
on hold if you have vad enabled.


On Mon, Mar 28, 2011 at 9:14 AM, Dan Lane <null at invalid.name> wrote:
> On Fri, Mar 25, 2011 at 8:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> other formats also with 120 ms ptime?
>
> Namely PCMA / PCMU. We've noticed that G722 is limited to 60ms and I
> don't seem to be able to get iLBC to work above 40ms.
>
>> You do run the risk of losing a significant amount of audio if you
>> drop even 1 120ms packet.
>> The other end would be starved for 5 intervals.
>
> Yes, but we're dealing with mobile endpoints behind very jittery connections.
>
>> you could try rtp_timer_name=none on the 120ms leg
>
> That fixed it... rtp_timer_name doesn't appear to be documented very
> well on the wiki so are there any negative side-effects to having this
> set to none? (I'll update the wiki with anything I learn as I'm doing
> this).
>
> Thanks :)
>
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-- 
Anthony Minessale II

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