If I only care about outbound audio, is there a way to force the audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there still this same issue?<div><br></div><div>--matt<br><br><div class="gmail_quote">
On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker <span dir="ltr"><<a href="mailto:rupa@rupa.com">rupa@rupa.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes.<br>
<br><div class="gmail_quote"><div><div></div><div class="h5">On Wed, Dec 23, 2009 at 9:55 AM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div class="h5">
<div style="word-wrap:break-word">That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff.<div>
<br></div><font color="#888888"><div>/b</div></font><div><div></div><div><div><br><div><div>On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:</div><br><blockquote type="cite"><span style="border-collapse:separate;font-family:Helvetica;font-size:medium;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">I use the SIP Termination service from ezcall inc (<a href="http://grnvoip.com/" target="_blank">grnvoip.com</a>) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH.<div>
<br></div><div>The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error</div>
<div><br></div><div> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20</div><div><br></div><div>I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem.</div>
<div><br></div><div>but now I'm getting a </div><div><br></div><div> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10</div><div><br></div><div>error and in some sessions the audio is playing back too fast (at 3x the speed).</div>
<div><br></div><div>Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds?</div><div><br></div><div><br></div><div>--matt</div></span></blockquote>
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