[asterisk-dev] Codec Negotiation for Asterisk-1.2.9.1

[Chan Kwang Mien]

Hi,

I tried the patch for codec negotiation for Asterisk-1.2.9.1
from http://unofficial.portaone.com/~bamby/public/

My test-bed settings are 

sip1 <--> Asterisk <--> sip2

sip1 configured to use g.729 and g.711u
sip2 configured to use g.729

In sip.conf,

[sip1]
allow = g729
allow = ulaw

[sip2]
allow = g729

sip1 is not successful in establishing a call to sip2.  

>From SIP Message Debug Logs below, it seems that Asterisk sent the
codecs : g.711u, gsm, g.711a in the SDP to sip2, although the allowed
codec for sip2 is g.729. Hence, sip2 returned "Unsupported Media Type"
message to Asterisk.

Has anyone tried this patch ? or is there a latest codec negotiation
patch for Asterisk-1.2.9.1 ?

Thank you.

regards,
Kwang Mien




SIP MESSAGE LOGS
================ 

We're at 192.168.100.143 port 13552
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (NAT) to 192.168.100.113:5060:
INVITE sip:2003 at 192.168.100.113:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.143:5060;branch=z9hG4bK781e611f;rport
From: "2006" <sip:2006 at 192.168.100.143>;tag=as27fbdd97
To: <sip:2003 at 192.168.100.113:5060>
Contact: <sip:2006 at 192.168.100.143>
Call-ID: 76018af3574d06fb2adac6ac24659fc1 at 192.168.100.143
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Aug 2006 08:15:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 25287 25287 IN IP4 192.168.100.143
s=session
c=IN IP4 192.168.100.143
t=0 0
m=audio 13552 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 2003
Aug 10 16:15:59 WARNING[25441]: channel.c:2320 set_format: Unable to
find a codec translation path from 0x100 (g729) to ulaw
Aug 10 16:15:59 WARNING[25441]: channel.c:2320 set_format: Unable to
find a codec translation path from 0x100 (g729) to ulaw

<-- SIP read from 192.168.100.113:5060:
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 192.168.100.143:5060;branch=z9hG4bK781e611f;rport
Call-ID: 76018af3574d06fb2adac6ac24659fc1 at 192.168.100.143
CSeq: 102 INVITE
From: "2006" <sip:2006 at 192.168.100.143>;tag=as27fbdd97
To: <sip:2003 at 192.168.100.113:5060>;tag=zfjaOVZY51SKgQKx
Contact: <sip:2003 at 192.168.100.113:5060>
Content-Length: 0


--- (8 headers 0 lines)---
    -- Got SIP response 415 "Unsupported Media Type" back from
192.168.100.113
Transmitting (NAT) to 192.168.100.113:5060:
ACK sip:2003 at 192.168.100.113:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.143:5060;branch=z9hG4bK781e611f;rport
From: "2006" <sip:2006 at 192.168.100.143>;tag=as27fbdd97
To: <sip:2003 at 192.168.100.113:5060>;tag=zfjaOVZY51SKgQKx
Contact: <sip:2006 at 192.168.100.143>
Call-ID: 76018af3574d06fb2adac6ac24659fc1 at 192.168.100.143
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0