[Freeswitch-users] Conference delay increasing over time

[Erik M. Devane - Comms Guy]

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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