[Freeswitch-users] Conference delay increasing over time

[Anthony Minessale]

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 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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>>
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>
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-- 
Anthony Minessale II

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