[Asterisk-Users] realtime audio for asterisk using jack

[Matt Riddell]

Esben Stien wrote:
> Matt Riddell <matt.riddell at sineapps.com> writes:
> 
> 
>>>Any plans for asterisk to support jack for realtime audio?,
>>
>>So that you can have a phone on the console of Asterisk?
> 
> Yes.

And what difference is 30ms going to make?  Bearing in mind that reverb 
mixes with original sound (to our ears) if the predelay is less than 
about 30ms...

:-)

BTW: I'm ZX81 - we talked yesterday on IRC until you mentioned the audio 
software you're using under LINUX (I quickly ran off to check them out).

Also, if you're comparing it to one of the iaxclient phones then you 
have to remember that under Windows they use 400ms delay - although DIAX 
has just made this configurable - I have mine set to 60ms.

So, the gist of what I'm saying is that on a local call (I.E. PSTN) the 
delay (say 50ms) would be totally unnoticeable and would probably be 
more apparent as side tone.

On a VOIP call (say 300ms) the 50ms is really not going to make that 
much difference.

I think the reason you want it is because you feel that chan_oss and 
chan_alsa are creating a half second delay.  I don't see that here.  Is 
it possible that JACK creates an emulated alsa/oss layer for non JACK 
connections?

-- 
Cheers,

Matt Riddell
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