[Freeswitch-users] Conference delay increasing over time

[Erik M. Devane - Comms Guy]

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 gmail.com> wrote:
>
> I have a very well specified Windows Server connected to sound card input
> via mod_portaudio.
>
> I want multiple callers to be able to listen (only) to the incoming feed,
> so I set up dynamic conferences, with the dialplan flags setting mute. An
> autocall bridges the portaudio.
>
> My assumption is that this is the best way. I am happy to be wrong if
> there's a better plan.
>
> The problem is that delay builds up, seconds every minute on the
> conference.
>
> I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
>
> Any other suggestions I could try? Would CentOS have better timing?
>
> I don't need callers to hear each other, but I'm sure that I'm missing
> something.
>
> Any thoughts?
>
> Erik
>
>
>
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> --
> Anthony Minessale II
>
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