[asterisk-bugs] [Asterisk 0010500]: [patch] teach chan_iax2 to offer the calling channel's codec first, like chan_sip does it

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10500 
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Reported By:                stevedavies
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10500
Category:                   Channels/chan_iax2
Reproducibility:            N/A
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 80048 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-20-2007 11:24 CDT
Last Modified:              08-20-2007 13:36 CDT
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Summary:                    [patch] teach chan_iax2 to offer the calling
channel's codec first, like chan_sip does it
Description: 
We discussed this issue on asterisk-dev under the title "IAX codec
selection".

Currently, outgoing IAX NEW packets contain two codec-related IEs. 
"FORMAT" will contain the format bits for the calling channel.  CODEC_PREFS
is a list of codecs allowed, taken as-is from the allow= of the peer.

Now it is preferable not to transcode - so it makes sense that the calling
channel's codec should be moved first in the CODEC_PREFS list (provided
that it is one of the allowed codecs).  

This is what chan_sip does.

So this patch brings that behaviour to IAX.

As a demonstration of the issue, here's a peer:

[xxxx]
type=friend
host=xxx.co.za
username=us
secret=ourpass
trunk=no
notransfer=yes
transfer=no
context=from-peer-xxxx
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=ilbc
allow=gsm
allow=speex
qualify=500
qualifysmoothing=yes

Now we send a call to this peer from a phone using iLBC.  chan_iax2 as was
sends this NEW:

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass:
NEW
Timestamp: 00017ms  SCall: 00001  DCall: 00000 [A.B.C.D:4569]
VERSION         : 2
CALLED NUMBER   : 082XXXXXXX
CODEC_PREFS     : (alaw|ulaw|g729|ilbc|gsm|speex)
CALLING NUMBER  : 021XXXXXXX
CALLING PRESNTN : 35
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME    : Steve Davies
LANGUAGE        : en
USERNAME        : us
FORMAT          : 1024
CAPABILITY      : 59150
ADSICPE         : 2
DATE TIME       : 2007-08-15  10:18:02

So even though ILBC is the current codec, the CODEC_PREFS has ALAW first.

After my patch, the NEW is:

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: NEW
   
Timestamp: 00016ms  SCall: 00012  DCall: 00000 [A.B.C.D:4569]
VERSION         : 2
CALLED NUMBER   : 082XXXXXXX
CODEC_PREFS     : (ilbc|alaw|ulaw|g729|gsm|speex)
CALLING NUMBER  : 021XXXXXXX
CALLING PRESNTN : 35
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME    : Steve Davies
LANGUAGE        : en
USERNAME        : us
FORMAT          : 1024
CAPABILITY      : 59150
ADSICPE         : 2
DATE TIME       : 2007-08-20  18:37:16


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---------------------------------------------------------------------- 
 pj - 08-20-07 13:36  
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this will work automaticaly? I think in SIP case, to avoid unneeded
transcoding, is needed to set some variable like in SIP_CODEC before dial
command... 
it will be great if asterisk now handle codecs negotiation better to avoid
transcoding in full voice path, ie. even in case crossing multiple
asterisks... 

Issue History 
Date Modified   Username       Field                    Change               
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08-20-07 13:36  pj             Note Added: 0069103                          
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