[asterisk-users] SIP and codec negotiation

[Steve Davies]

Hi,

If I am expecting too much here, please just tell me so, but I was
under the impression that this was put into 1.6.x

I have 2 types of SIP devices. For argument's sake, let us say that
one type of device can talk G722 and ALAW, and the other only talks
ALAW. I have directmedia=yes.

Calls originated from ALAW only devices work great.
Calls from G722 to G722 devices work great.

...but the G722 to ALAW calls do not work. I can see from the SIP
trace that this is because Asterisk makes no attempt to modify the
codecs in its directmedia re-INVITE packets to ensure that the 2
parties can talk, so you end up with an asymmetric codec stream
between the handsets, which results in silence both ways. I would
expect Asterisk to either determine that there are no common codecs,
and do an implicit "directmedia=no" for the remainder of the call, or
to only send the list of common codecs to each party in the
re-INVITE's SDP (There is a room for a per-device
"can_change_codec=bool" parameter in there too I think).

For 1.4 there was a popular codec negotiation patch which I believe
fixed this. Is this not in 1.6? Am I missing something else perhaps?

Thanks for any pointers.

Regards,
Steve