[asterisk-dev] Multiple codecs on SDP answer if SIP_CODEC variable is used

[Saúl Ibarra Corretgé]

Hi,

I'm experiencing some unexpected (at least to me) behavior when the
SIP_CODEC variable is set and I'd like some input before posting an
issue.

Test case: assume we have the following configuration:

[telf](!)
type=friend
secret=1234
disallow=all
allow=g722
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
context=saghul-test
host=dynamic

and

[saghul-test]
exten => 12346,1,NoOp()
same => n,Set(__SIP_CODEC=g722)
same => n,Answer
same => n,Wait(1)
same => n,MusicOnHold(default)

If a user dials 12346 *with* G722 on his SDP offer the answer
generated by Asterisk contains a single codec: G722, all ok:

v=0
o=root 542332501 542332501 IN IP4 192.168.99.53
s=Asterisk PBX SVN-branch-1.8-r294084
c=IN IP4 192.168.99.53
t=0 0
m=audio 15168 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

However, if the SDP offer *doesn't contain* G722, Asterisk prints a
log line saying that both parties don't support G722 so it won't be
used. That's fine, but the SDP answer generated contains all the
codecs that were offered and are allowed in the peer definition:

v=0
o=root 911458123 911458123 IN IP4 192.168.99.53
s=Asterisk PBX SVN-branch-1.8-r294084
c=IN IP4 192.168.99.53
t=0 0
m=audio 16716 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Is this intended? I would expect the answer to contain a single codec,
it being the next one that could possibly be used. Is this a bug or a
feature? :-)


Thanks and regards,

-- 
/Saúl
http://saghul.net | http://sipdoc.net