|Summary:||ASTERISK-02207: [request] make generators work even if no received audio available|
|Date Opened:||2004-08-09 10:26:11||Date Closed:||2011-06-07 14:05:04|
|Description:||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.
If we really really want to get this sorted could my company offer a bounty? or would some work from Digium be required to get this fixed?
Buying an E1 card would be our last option.
|Comments:||By: Olle Johansson (oej) 2004-08-09 10:40:04|
Announce a bounty on the wiki or find a consultant on the list on the wiki :-)
By: Olle Johansson (oej) 2004-08-09 11:56:24
Please add bounty amount and contact details.
By: Olle Johansson (oej) 2004-08-09 12:23:59
Moved to the wiki.