|Summary:||ASTERISK-17109: Codec negotiation algorithm|
|Reporter:||Marco Marzetti (manzo_zeti)||Labels:|
|Date Opened:||2010-12-15 07:19:26.000-0600||Date Closed:||2011-06-07 14:01:02|
|Description:||According to the provided documentation ( sip.conf ), when Asterisk is placing a call the CODEC used will be the first CODEC in the allowed CODEC's that the caller indicates that it supports.|
So if, for instance, we have the following setup:
RTP stream from caller to called would be transcoded from $WATHEVER to G.729.
Such behavior is sub-optimal when $CUSTOMERS devices come in the scene since
we can't be sure that our configured codec preferences are the same as theirs.
I think Asterisk should place call using the first CODEC sent by the caller in the INVITE message that [caller] indicates that it supports.
|Comments:||By: Igor Zamocky (dzajro) 2010-12-15 11:33:05.000-0600|
There are many scenarios benefiting from current mechanism. But maybe it could be configuration related. Something like "honor host preferences" vs. "honor my preferences".
By: Paul Belanger (pabelanger) 2010-12-15 14:13:28.000-0600
Features requests are no longer submitted to or accepted through the issue tracker. Features requests are openly discussed on the mailing lists  and Asterisk IRC channels and made note of by Bug Marshals.