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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