Wow... Thinking about this timer setting and about how it converted send()/recv() from non-blocking to blocking, I straced freeswitch when it was supposed to be idle. It never pauses! It keeps going in and out of select() every millisecond! Why??
------ Original Message ------ Received: Tue, 01 Dec 2009 08:31:46 PM EST From: [email protected] To: <[email protected]> Subject: Re: [Freeswitch-users] Choppy sound with PCMU > 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 <[email protected]> > To: [email protected] > 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 <[email protected]> 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. > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > [email protected] > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:[email protected] <msn%[email protected]> > > > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:[email protected] <sip%[email protected]> > > iax:[email protected]/888 > > > googletalk:[email protected]<googletalk%3aconf%[email protected]> > > pstn:213-799-1400 > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > [email protected] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
