[Freeswitch-users] Conference delay increasing over time

[covici at ccs.covici.com]

I am not seeing this here at all using real hardware and latest git as
of a few days ago.  I was getting a year maybe or so ago (not as bad as
yours),  but it went away and I never figured out why.

Erik M. Devane - Comms Guy <emdevane at gmail.com> wrote:

> Increasing the conference internal to 10 to 20 to 200 had no detectable
> effect.
> 
> In order to improve timing, I rebuilt the server as Centos 6.4, building
> from the tarball (git had some proxy-related issues).
> 
> [New issue on Linux]
> I haven't been able to get mod_portaudio to give me more than two mono
> inputs per M-Audio Delta 1010 card (though alsamixer and pa_devs seem
> perfectly happy).
> 
> 
> [Back to the original issue]
> I reverted all RTP timer, flush, and other changes, back very close to the
> default configuration.
> 
> Still I have a delay, still increases seconds per minute.
> 
> I explored moh but couldn't figure out how to have multiple playlists. If
> this is possible, Id love to know how!
> 
> My best solution so far has been to have the first call bridge portaudio
> and subsequent calls eavesdrop on that first bridged call. Latency is
> almost as good as a single bridged call, and I have tested six
> eavesdroppers with no ill effects.
> 
> Right now I'm going to:
> 
> 1) Work out how to mute external caller's audio from the eavesdrop.
> 
> 2) Learn lua enough to roll an eavesdropper onto the bridge when the first
> caller hangs up.
> 
> It's not a satisfactory solution, but I'm out of ideas for the conference
> lag unless anyone else can help, and I don't know how to do multiple MOH
> instances.
> 
> Any help would be most appreciated - either on conference lag or moh or
> eavesdrop mute.
> 
> Erik
> 
> On Tuesday, November 12, 2013, Anthony Minessale wrote:
> 
> > https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc
> >
> > Its just setting up mod_local_stream to point at a dir with a file with a
> > .loc extension that has the string of the pa url in it.
> >
> > You should probably not change any of those params away from default.  I
> > would recommend putting those back to where they belong.
> >
> > If anything it could worsen your problem.  What about the interval?  DId
> > you try increasing that?
> >
> >
> >
> >
> >
> >
> >
> >
> > On Mon, Nov 11, 2013 at 11:00 PM, Erik M. Devane - Comms Guy <
> > emdevane at gmail.com> wrote:
> >
> > I've been working at this all day, with no joy.
> >
> > I thought that antivirus software was to blame, but that was a dead end.
> >
> > Does anyone have an example of running the .loc local_stream approach?
> >
> >
> > On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:
> >
> > OK, so I've had another chance to work with the settings following your
> > suggestion.
> >
> > Setup - mixing console sending identical channels to sixteen channels on
> > two M-Audio Delta 1010 cards.
> > SIP trunk from Cisco system, testing using AT&T phone to public phone
> > number. CODEC is PCMU/8000.
> >
> > Everything seems perfect when just doing a simple bridge:
> > <action application="bridge"
> > data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/>
> > I can listen to it for hours, with no issues or delay.
> >
> >
> > Problems start when creating a default conference used, by dialplan:
> >
> > <action application="conference_set_auto_outcall"
> > data="portaudio/endpoint/MAUDIO-11"/>
> > <action application="conference" data="$1-${domain_name}@
> > ${use_profile}++flags{mute}""/>
> >
> > Test - call both bridged extension and conference, and listen. After a
> > minute, there is definite delay in the conference. After three minutes,
> > there is a second delay. After ten minutes, the audio is so far behind it
> > is unusable.
> >
> > Setup:
> >
> > Conference - default, energy level: 100, waste, all callers set to mute.
> >
> > External.xml used for profile, changes to the default:
> >
> > <param name="rtp-timer-name" value="none"/>
> > <param name="rtp-autoflush-during-bridge" value="true"/>
> > <param name="rtp-autoflush" value="true"/>
> >
> > Setting PortAudio rate to 8000, using default conference rate 8000, audio
> > sounds the same, and no issues over bridge. Conference still has delay..
> >
> > Setting PortAudio rate to 48000, upping default conference rate to 48000,
> > audio sounds the same, and no issues over bridge. Conference still has
> > delay.
> >
> > Any other suggestions would be gladly received. I couldn't locate many
> > examples of using soundcards as an MOH loc and streaming that, so that is
> > another avenue to try, if anyone has any hints.
> >
> > Thank you for your earlier suggestions - that I can stream sixteen
> > channels to the outside world reliably, with sensible configuration
> > options, is an outstanding achievement by the developers. Now if I could
> > have multiple callers receive the same audio...
> >
> >
> >
> >
> > On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale <
> > anthony.minessale at gmail.com> wrote:
> >
> > Are you running the pa and the conference both at a high rate?
> > Some soundcards do bad at slower rates since its emulated.  Its most
> > likely the timing on the soundcard over anything else.
> >
> >
> > On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy <
> > emdevane at gmail.com> wrote:
> >
> > No, I hadn't - that sounds good. I'm using the new(ish) PortAudio
> > shstreams endpoints and have been trying to find examples of the .loc
> > approach with multiple soundcards.
> >
> > Any guidance welcomed!
> >
> > Does anyone have any thoughts on why conferences would be slowing down?
> >
> > On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale <
> > anthony.minessale at gmail.com> wrote:
> >
> > have you seen mod_portaudio_stream you can use that in a .loc file
> > together with mod_local_stream for static muxing and just play the
> > localstream as a file in your dialplan
> >
> >
> > On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy <emdevane at g
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <javascript:_e({}, 'cvml',
> > 'sip%3A888 at conference.freeswitch.org');>
> > googletalk:conf+888 at conference.freeswitch.org <javascript:_e({}, 'cvml',
> > 'googletalk%3Aconf%2B888 at conference.freeswitch.org');>
> > pstn:+19193869900
> >
> 
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com