[asterisk-bugs] [Asterisk 0018476]: Codec negotiation algorithm

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=18476 
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Reported By:                manzo_zeti
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18476
Category:                   Codecs/General
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.15 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2010-12-15 07:19 CST
Last Modified:              2010-12-15 14:13 CST
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Summary:                    Codec negotiation algorithm
Description: 
According to the provided documentation ( sip.conf ), when Asterisk is
placing a call the CODEC used will be the first CODEC in the allowed
CODEC's that the caller indicates that it supports.

So if, for instance, we have the following setup:

[caller]
disallow=all
allow=g729
allow=alaw

[called]
disallow=all
allow=alaw
allow=gsm

RTP stream from caller to called would be transcoded from $WATHEVER to
G.729.
Such behavior is sub-optimal when $CUSTOMERS devices come in the scene
since 
we can't be sure that our configured codec preferences are the same as
theirs.

I think Asterisk should place call using the first CODEC sent by the
caller in the INVITE message that [caller] indicates that it supports.

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---------------------------------------------------------------------- 
 (0129623) pabelanger (administrator) - 2010-12-15 14:13
 https://issues.asterisk.org/view.php?id=18476#c129623 
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Features requests are no longer submitted to or accepted through the issue
tracker. Features requests are openly discussed on the mailing lists [1]
and Asterisk IRC channels and made note of by Bug Marshals.

[1] http://www.asterisk.org/support/mailing-lists 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-15 14:13 pabelanger     Note Added: 0129623                          
2010-12-15 14:13 pabelanger     Status                   new => closed       
2010-12-15 14:13 pabelanger     Resolution               open => no change
required
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