Is ptime adaptable? i won&#39;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?<div>
<br></div><div>Btw, everything works fine now, with 20ms, 160 bytes chunk and FS just play that smoothly. Thanks for all the pointer.</div><div><br></div><div>JK<br><br><div class="gmail_quote">On Thu, Aug 26, 2010 at 12:49 PM, Michael Jerris <span dir="ltr">&lt;<a href="mailto:mike@jerris.com">mike@jerris.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">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?<div>
<br></div><div>Mike</div><div><div></div><div class="h5"><div><br><div><div>On Aug 26, 2010, at 10:15 AM, Johny Kadarisman Kwan wrote:</div><br><blockquote type="cite">I adjust rtp timestamp to increment += 160, still no good audio. <div>
i took code that handle speex previously, timestamp was set to 320increment. seems working fine with speex/16k</div><div><br></div><div>still no good ulaw audio. I&#39;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)?</div>

<div><br></div><div>Thanks again.<br><br><div class="gmail_quote">On Thu, Aug 26, 2010 at 10:02 AM, Brian West <span dir="ltr">&lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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&#39;re still going to get the warning I&#39;m pretty sure.<br>


<font color="#888888"><br>
/b<br>
</font><div><br>
On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:<br>
<br>
&gt; Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes payload does eliminate warning message from FS.<br>
&gt; My audio doesn&#39;t work yet, problem must be something else. At least no more issues on rtp audio framing ;)<br>
&gt;<br>
&gt; Thanks,<br>
&gt; JK<br>
<br>
<br>
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