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

[Asterisk Bug Tracker]

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:                   block
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:              2010-03-15 11:36 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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 (0119390) jonaskellens (reporter) - 2010-03-15 11:36
 https://issues.asterisk.org/view.php?id=17020#c119390 
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When a G729-capable phone calls a non-capable phone through Asterisk then
there is always the following that happens :

INVITE from g729-capable phone to Asterisk results in :

Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x10a
(gsm|alaw|g729)/video=0x0 (nothing), combined - 0x10a (gsm|alaw|g729)

SDP in INVITE from Asterisk to non-capable phone :

Content-Type: application/sdp
Content-Length: 265

v=0
o=root 24971 24971 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 11580 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

And the result is :

<--- SIP read from 192.168.1.107:5060 --->
SIP/2.0 415 Unsupported Media Type


So my point is : why does Asterisk only present g729 as codec ?? Why not
present g729, alaw and gsm to the called peer ??

Then the peer can reply with codecs alaw & gsm in its SDP-body.

The moment I place the g729 as second codec in sip.conf, then the call
uses alaw for both channels. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-15 11:36 jonaskellens   Note Added: 0119390                          
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