Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just installing SVN trunk didn't fix it either, but setting that option afterwards surely did the trick. One thing I've noticed while staring at the console is that it *looks like* that w/o the new setting the stuttering happens when FS either re-registers itself with the provider or one of the SPA's port re-registers with FS. ------ Original Message ------ Received: Tue, 01 Dec 2009 05:33:26 PM EST From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Choppy sound with PCMU
linksys has had a bug for eons that can be fixed by setting the ptime (or rtp packet size in their terms) in it's firmware to .20 instead of .30
Asterisk does not use async RTP like we do so it's never a problem 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.
You should also test this on latest SVN trunk or wait for pre8
On Tue, Dec 1, 2009 at 3:52 PM, eaf <erandr-junk@usa.net> wrote:
I should also add, after browsing through some topics here, that my SIP provider sends 172-byte RTP frames, which is in accordance with ptime:20 that it gives to FreeSWITCH.
eaf wrote:
Hi,
I'm trying to migrate from Asterisk to FreeSWITCH (really like the way
how
it can be programmed), but ran into one issue with sound quality that I just cannot workaround by myself. I would describe the sound problem as being "choppy". From time to time small portions of the other party's voice are dropped, so the voice kind of stutters. This is not too bad, but is really noticeable, happens in every call and I don't experience the same with Asterisk running on the same box. I attached two files: freeswitch.wav and asterisk.mp3 to illustrate my point.
Issue completely goes away, if I set inbound-proxy-media to true.
The way how I test is to connect SPA-2000 via 10mbps LAN to the box directly exposed to internet, and then dial a toll-free via FutureNine
(a
SIP provider).
The codec in use is PCMU. Can't really try PCMA or anything else with this provider. Only PCMU. Tried to match ptime of provider (30) with ptime of the SPA, didn't get any improvement. Tried turning off recording, no change either.
What puzzles me is that even with greedy codec negotiations and with PCMU on both sides of FreeSWITCH, it's still saying that TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to illustrate.
The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that it's not a performance issue.
http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav
http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3
http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log
Tried both 1.0.4 and 1.0.5pre5. Same results.
What should I do next? Calls are consistently bad with FreeSWITCH, and consistently show no glitches with Asterisk.
-- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html Sent from the Freeswitch-users mailing list archive at Nabble.com.
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