[Freeswitch-dev] FS complained on RTP packet.

[Johny Kadarisman Kwan]

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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