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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