linksys has had a bug for eons that can be fixed by setting the ptime (or rtp packet size in their terms)<br>in it's firmware to .20 instead of .30<br><br>Asterisk does not use async RTP like we do so it's never a problem<br>
you can disable the timer by setting the channel var rtp_timer_name=none or sofia param rtp-timer-name to none in the sofia profile.<br><br>You should also test this on latest SVN trunk or wait for pre8<br><br><br><br><div class="gmail_quote">
On Tue, Dec 1, 2009 at 3:52 PM, eaf <span dir="ltr"><<a href="mailto:erandr-junk@usa.net">erandr-junk@usa.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
I should also add, after browsing through some topics here, that my SIP<br>
provider sends 172-byte RTP frames, which is in accordance with ptime:20<br>
that it gives to FreeSWITCH.<br>
<div><div></div><div class="h5"><br>
<br>
eaf wrote:<br>
><br>
> Hi,<br>
><br>
> I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how<br>
> it can be programmed), but ran into one issue with sound quality that I<br>
> just cannot workaround by myself. I would describe the sound problem as<br>
> being "choppy". From time to time small portions of the other party's<br>
> voice are dropped, so the voice kind of stutters. This is not too bad, but<br>
> is really noticeable, happens in every call and I don't experience the<br>
> same with Asterisk running on the same box. I attached two files:<br>
> freeswitch.wav and asterisk.mp3 to illustrate my point.<br>
><br>
> Issue completely goes away, if I set inbound-proxy-media to true.<br>
><br>
> The way how I test is to connect SPA-2000 via 10mbps LAN to the box<br>
> directly exposed to internet, and then dial a toll-free via FutureNine (a<br>
> SIP provider).<br>
><br>
> The codec in use is PCMU. Can't really try PCMA or anything else with this<br>
> provider. Only PCMU. Tried to match ptime of provider (30) with ptime of<br>
> the SPA, didn't get any improvement. Tried turning off recording, no<br>
> change either.<br>
><br>
> What puzzles me is that even with greedy codec negotiations and with PCMU<br>
> on both sides of FreeSWITCH, it's still saying that<br>
> TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to<br>
> illustrate.<br>
><br>
> The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800<br>
> with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that<br>
> it's not a performance issue.<br>
><br>
> <a href="http://old.nabble.com/file/p26594250/freeswitch.wav" target="_blank">http://old.nabble.com/file/p26594250/freeswitch.wav</a> freeswitch.wav<br>
><br>
> <a href="http://old.nabble.com/file/p26594250/asterisk.mp3" target="_blank">http://old.nabble.com/file/p26594250/asterisk.mp3</a> asterisk.mp3<br>
><br>
> <a href="http://old.nabble.com/file/p26594250/freeswitch.log" target="_blank">http://old.nabble.com/file/p26594250/freeswitch.log</a> freeswitch.log<br>
><br>
> Tried both 1.0.4 and 1.0.5pre5. Same results.<br>
><br>
> What should I do next? Calls are consistently bad with FreeSWITCH, and<br>
> consistently show no glitches with Asterisk.<br>
><br>
><br>
<br>
--<br>
</div></div>View this message in context: <a href="http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html" target="_blank">http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html</a><br>
<div><div></div><div class="h5">Sent from the Freeswitch-users mailing list archive at Nabble.com.<br>
<br>
<br>
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