Low latency audio distribution


you may not believe it but i have a problem :wink:

The scenario is as follows: There is a TV running a favourite News Channel with Stock Information etc… The TV is muted so that nobody is disturbed by it. When someone wants to listen to the program, he dials a certain number on his IP Phone and can hear the audio part via headset.

My idea was to use the MoH feature from Asterisk and grab the music from the Line-In of the soundcard. I built and tested it using this howto: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf#Usingasoundcardasthesource

Added these two lines to the extensions.conf:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()

It is working but there is a huge delay between audio an video at least 4 to 5 seconds. Do you have any idea where this delay comes from and how to get rid of it? CPU utilisation is not an issue it’s always at 0-1%. Is there any better/other way to do this?

regards Cabal

i butchered some BudgeTone 102’s recently for overhead paging, so this might work …

modify the BudgeTone so that the handset mic is linked to a new input jack socket, and this is “fed” by the tv output. set the phone to autoanswer and call it before transfering it into a meetme room.

then everyone else can call into the meetme (maybe mute them too). the only problem i can see is if Asterisk is restarted or the call from the phone is dropped. but you could probably automate that anyway.


okay the meetme approach sounds interesting indeed. I think about it but would like to modify the setup. What do you think about using a Softphone which at startup automatically calls into the meetme room and uses the line-in as audio source. This would run on the same machine as the asterisk does therefore it has to be a commandline application any recommendations anyone?

regards Cabal

if you’re using the Asterisk machine you could just use the Console channel … voip-info.org/wiki/index.php … ps+console

i think console+meetme is your best bet. It should be lag-free and easy to use.

cheap grandstream phone would be good too, that way you could put the phone next to the TV…


thanks for your help it’s almost done. I have loaded the chan_alsa.so now if you type “dial 200” on the asterisk CLI, the phone with number 200 is ringing. If you answer the call you can hear the sound of the TV with a much smaller delay (<1/2second). This is much faster and the audio quality is significantly better then before.

Meetme is also working but i have no idea howto add the ‘ALSA/default’ automatically at startup to the meetme conference. Maybe you can help me once more?

regards Cabal

On the console, dial the exten that puts it in the conference room. Have everybody else dial something which puts them in, but muted.


is it possible to have multiple console channels? We would like to add another conference room, in order to distribute further audio streams.

Regards Cabal