[Asterisk-Dev] sdp: asterisk responds with codecs not in INVITE

[Alex Zeffertt]

Hi all,

I've spotted what I think is a bug in chan_sip.c.  I'm new to this code
so please forgive me if I'm talking nonsense.

Anyway, the problem is that when asterisk responds to a SIP INVITE with
a SIP 200 OK, the attached SDP payload may contain codecs not present
in the original INVITE.  I believe that it should instead respond with
the intersection of the codecs it understands and the codecs offered in
the original INVITE.

I think the problem is in check_user_full() in chan_sip.c.  Once
this has confirmed the user exists and the password is correct it
resets the jointcapability to the capabilties listed in sip.conf:

				p->jointcapability = user->capability;

The effect of this is to ignore what was offered in the INVITE.

Can anybody suggest what I should do?  Will it have any negative side
effects just to comment out this line?


-- 
Alex Zeffertt
Software Engineer
Cambridge Broadband Ltd.
http://www.cambridgebroadband.com