[Freeswitch-dev] FS complained on RTP packet.

[Johny Kadarisman Kwan]

Is ptime adaptable? i won't know what is the rate at the beginning of
sip/sdp negotiation. But it possible to calculate how much audio once i
process the upstream audio. So, is it possible to change the ptime while
call in progress phase?

Btw, everything works fine now, with 20ms, 160 bytes chunk and FS just play
that smoothly. Thanks for all the pointer.

JK

On Thu, Aug 26, 2010 at 12:49 PM, Michael Jerris <mike at jerris.com> wrote:

> If you are already getting them in larger chunks, you might as well pass
> along the packet size you get instead of breaking them up, just make sure to
> set the ptime correct.  What is the source of the audio?
>
> Mike
>
> On Aug 26, 2010, at 10:15 AM, Johny Kadarisman Kwan wrote:
>
> I adjust rtp timestamp to increment += 160, still no good audio.
> i took code that handle speex previously, timestamp was set to
> 320increment. seems working fine with speex/16k
>
> still no good ulaw audio. I'm converting up stream audio that sent to me in
> a large chunk, do some processing and now breaking up into smaller rtp
> chunk. do i have to limit the rtp paket rate to freeswitch (ie. 20ms delay
> between them)?
>
> Thanks again.
>
> On Thu, Aug 26, 2010 at 10:02 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Its could be your timestamps too... how many are you incrementing on each
>> time stamp?  If you lie about time timestamps say send timestamps that jump
>> by 320 but only send 160 byte payload you're still going to get the warning
>> I'm pretty sure.
>>
>> /b
>>
>> On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:
>>
>> > Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes
>> payload does eliminate warning message from FS.
>> > My audio doesn't work yet, problem must be something else. At least no
>> more issues on rtp audio framing ;)
>> >
>> > Thanks,
>> > JK
>>
>>
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