[asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11376 
====================================================================== 
Reported By:                lasse
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11376
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.14  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             11-26-2007 07:51 CST
Last Modified:              11-27-2007 01:50 CST
====================================================================== 
Summary:                    Codec negotiation results in asterisk sending
unsupported codec
Description: 
SIP Phone client that  supports only iLBC
Asterisk configured to support alaw, ilbc in that order (sip.conf).
PSTN provider supports all kinds of codecs

Call from PSTN to sip phone works fine (PSTN -> alaw -> Asterisk -> ilbc
-> SIP client)

BUT if the order of the codecs in sip.conf is ilbc, alaw asterisk will
send ilbc to PSTN provider and alaw to SIP client (allthough it indicates
it only support ilbc in SDP) resulting in a failed call.
====================================================================== 

---------------------------------------------------------------------- 
 lasse - 11-27-07 01:50  
---------------------------------------------------------------------- 
Unless i misunderstood something, the sdp from the client is sent in this
message:

asl004*CLI> 
<--- SIP read from 1.2.3.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK7e06ef7f;rport=5070
From: "Owner of number" <sip:+46987654321 at 1.2.3.4:5070>;tag=as3300c472
To: "unknown" <sip:46123456789 at 1.2.3.4>;tag=ae71559a12
Contact: <sip:46123456789 at 213.50.52.7:63066>
Call-ID: 197def4c6c6d17414029b0ef196ad339 at 1.2.3.4
CSeq: 102 INVITE
Content-Length: 249
Content-Type: application/sdp
Record-Route: <sip:1.2.3.4;lr;ftag=as3300c472>
Server: SJphone/1.65.377a (SJ Labs)
Supported: replaces,norefersub,timer

v=0
o=- 3405072225 3405072225 IN IP4 10.10.4.18
s=SJphone
c=IN IP4 1.2.3.4
t=0 0
m=audio 60202 RTP/AVP 97 101
c=IN IP4 1.2.3.4
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 1.2.3.4:60202
Found description format iLBC for ID 97
Found description format telephone-event for ID 101
Capabilities: us - 0x408 (alaw|ilbc), peer - audio=0x400 (ilbc)/video=0x0
(nothing), combined - 0x400 (ilbc)

And asterisk seems to conclude iLBC is the combined capability but then
proceeds to send alaw:
...
Sent RTP P2P packet to 1.2.3.4:60202 (type 08, len 000160)
... 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-27-07 01:50  lasse          Note Added: 0074393                          
======================================================================