[Asterisk-Dev] Re: How to implement an audio delay?

[Tony Mountifield]

In article <41EFBEB7.1040801 at stevek.com>, Steve Kann <stevek at stevek.com> wrote:
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> 
> Peter Svensson wrote:
> 
> >Perhaps the new jitter buffer implementation could be tweaked to insert 
> >that large a delay. If you search the mailing lists and the bug tracker 
> >you can get in touch with the people developing the new jitter buffer.
> >
> That would be me -- hopefully other people will join in at some point :)
> 
> The jitterbuffer has a buffer like this that you could use for this, and 
> you could certainly hack the jitterbuffer so that it thinks there's any 
> number of milliseconds of jitter, and it would delay by that amount..

I guess this would only be applicable to channels that use a jitter buffer,
e.g. SIP or IAX. Or is the jitter buffer used more generally within Asterisk?

> But, I think that, for this application, I'd write an app which takes a 
> call, makes a call, and acts as the "bridge" between them, with the 
> appropriate buffering of frames. You could call this app_delay, and have 
> a dialplan entry like Delay(<destination>, delay), which would make a 
> call to <destination> and add the delay specified in delay..

That is the way I was thinking of doing it, but it seemed like I would
be duplicating a lot of the functionality of app_dial. Unless there
is a way I can somehow pipeline into the existing app_dial. There are
large parts of the Asterisk code that I have yet to become familiar with!

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org