Matt Riddell <[EMAIL PROTECTED]> 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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