On Tue, Dec 1, 2009 at 9:19 PM, <erandr-junk@usa.net> wrote:

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: erandr-junk@usa.net
To: <freeswitch-users@lists.freeswitch.org>
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 <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.
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users@lists.freeswitch.org
> > > 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com>
> >
>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org>
> > iax:guest@conference.freeswitch.org/888
> >
>
googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org>
> > pstn:213-799-1400
> >
>
> > _______________________________________________
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> >
>
>
>
>
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