[Asterisk-Users] Codec matching weirdness

Dustin Goodwin Wed, 14 Jan 2004 23:01:02 -0800

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


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[Asterisk-Users] Codec matching weirdness

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