I am experiencing a problem that from list archive it appears others are
running into. When I dial from Cisco 7960 via the * to Free World Dialup
destinations that supports G.729 the call fails. The major error from
the debug log is
Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format:
Unable to find a path from G729A to ULAW
Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format:
Unable to find a path from ULAW to G729A
So I compared the SDP info coming from the 7960, sent out from * and
returning from the destination system and I have included them below.
Question 1: Why is * sending out SDP info that is different from the SDP
info contained in original SDP from the phone?
Question 2: Is there a config option to force * to just passthrough the
codec list sent by the 7960 in the invite?
Question 3: What are SDP codec matching rules for SIP endpoints? How do
they decide on common codec. Comparing the SDP sent and receive all
systems claim support for 3 common codecs:
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
Now of course each device specified these 3 codecs in a different order.
Under normal circumstances I feel this call should complete why is *
claiming a codec mismatch?
- Dustin -
From phone
v=0
o=Cisco-SIPUA 5892 12461 IN IP4 192.168.68.12
s=SIP Call
c=IN IP4 192.168.68.12
t=0 0
m=audio 18114 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Sent to remote server by *
v=0
o=root 4205 4205 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 16798 RTP/AVP 4 3 0 8 2 5 10 7 18 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 192.246.69.223:5060
Received from remote server
v=0
o=root 9755 9756 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 10066 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16