[asterisk-bugs] [Asterisk 0017020]: SIP response 415 "Unsupported Media Type" when using G729

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Aug 30 12:36:47 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17020 
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Reported By:                jonaskellens
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17020
Category:                   Codecs/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           Older 1.4 - please test a newer version 
JIRA:                       SWP-1083 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-03-13 10:23 CST
Last Modified:              2010-08-30 12:36 CDT
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Summary:                    SIP response 415 "Unsupported Media Type" when using
G729
Description: 
Using Asterisk 1.4.25.1

Grandstream IP-phone has codecs G729, alaw, GSM.
Zoiper softphone has codecs alaw, GSM.

sip.conf peer definition grandstream :
disallow : all
allow : G729;alaw;GSM

sip.conf peer definition zoiper :
disallow : all
allow : alaw;GSM

When calling from zoiper softphone to Grandstream, the codec used is
alaw.

When calling from Grandstream to Zoiper, the call fails with SIP response
415 "Unsupported Media Type".

Why is it that Asterisk does not negotiates about the alaw-codec between
the Grandstream, Asterisk itself and the Zoiper softphone.

Is it normal behaviour that when one of the 2 SIP endpoints does not have
a G.729 license and Asterisk does not hold the license to translate from
G729 codec to another, the call fails ?
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---------------------------------------------------------------------- 
 (0126439) dvossel (administrator) - 2010-08-30 12:36
 https://issues.asterisk.org/view.php?id=17020#c126439 
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Are you using the 'directmedia' or 'canreinvite' option in sip.conf? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-30 12:36 dvossel        Note Added: 0126439                          
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