[Freeswitch-users] Choppy sound with PCMU

Anthony Minessale anthony.minessale at gmail.com
Wed Dec 2 10:32:21 PST 2009 

idle is a 4 letter word to a realtime application.

The core keeps a single high-priority thread to keep 1ms timing and expands
that broadcasting
to hundreds or thousand of threads who need accurate timing.

Your choppy audio is caused by linksys lying about the packet len that it's
using and we set our timer
to the wrong speed.


On Tue, Dec 1, 2009 at 9:19 PM, <erandr-junk at 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 at usa.net
> To: <freeswitch-users at 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 at gmail.com>
> > To: freeswitch-users at 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 at 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.wavfreeswitch.wav
> > > > >
> > > > >  http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3
> > > > >
> > > > >  http://old.nabble.com/file/p26594250/freeswitch.logfreeswitch.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
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> > > >
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> > > > 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 at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > >
> >
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> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > > IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
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> > >
> >
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> >
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> > >
> >
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> >
> >
> >
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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