[Asterisk-Users] New feature request

[Bart Coppens]

Dear all,

This feature request is derived from Bug ID 2206

Currently, Asterisk is using the timing of the input stream to reproduce the 
output stream. This means that when no RTP streams are being sent from the 
peer Endpoint/GW, Asterisk is unable not generate audio.  This 
approach/limitation can lead to "one way speech" conditions:

Some devices don't generate audio until the answer supervision is received 
from the called. For all these scenarios, no ringback can be presented to 
the calling party.

In cases where the endpoints are using silence compression, the audio from 
asterisk is chopped.

It would be much better to generate audio, even if no RTP is received at 
all. The clocking should than be taken from an internal timing mechanism 
that keeps track of the synchronization. A configuration option should exist 
to choose on the method.


Is anybody else interested in such feature request?

Cheers,
Bart

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