[asterisk-bugs] [Asterisk 0010604]: Codec options in gtalk.conf not respected

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10604 
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Reported By:                keepitcool
Assigned To:                phsultan
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Project:                    Asterisk
Issue ID:                   10604
Category:                   Channels/chan_gtalk
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-30-2007 08:00 CDT
Last Modified:              10-04-2007 15:54 CDT
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Summary:                    Codec options in gtalk.conf not respected
Description: 
When you have in gtalk.conf the following configuration:
[buddy]
disallow=all
allow=gsm
...

The gtalk connection is established even if the gtalk client do not
support the GSM codec, and the connection/channel appears established with
a different codec that it is not allowed (like for example the slin
codec).

On the other side, a similar configuration with IAX works as it is
supposed to.
When I have a configuration like this on my iax.conf :
disallow=all
allow=gsm
...

And when my remote client do not have GSM as one of the possible codecs,
the connect attempt is rejected and I receive the following message in my
zoiper iax client : “bearercapability notavail”

And on the asterisk server side I have the following message:
[Aug 30 11:40:50] NOTICE[1358]: chan_iax2.c:7645 socket_process: Rejected
connect attempt from XX.XX.XX.XX, requested/capability 0x200/0x60c
incompatible with our capability 0xe002.

Isn’t it suppose to work the same way with the gtalk ?

Testing components:
- Fedora core 3 + Asterisk server 1.4.11 (no zaptel, no libpri) + iksemel
1.3
- Google Talk client 1.0.0.104
- With the following patchs applied:
http://bugs.digium.com/view.php?id=10509 
http://bugs.digium.com/view.php?id=10548 (branch-1.4-10548-3.diff)

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---------------------------------------------------------------------- 
 phsultan - 10-04-07 15:54  
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How are you placing the call? GoogleTalk -> Asterisk -> IAX or the opposite
way? Please try both call configurations.

Also, an console debug output (along with an 'rtp debug' output) will help
solve the half way audio problem you're experiencing. I am not able to
reproduce it, thanks!

I'll try to answer your other questions :
- it is currently not possible to have the audio streams bypass Asterisk
when a GoogleTalk client is involved in the call ;
- the best places to start discussions about new features or configuration
issues are the Asterisk user/developer mailing lists, and the IRC channel. 

Issue History 
Date Modified   Username       Field                    Change               
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10-04-07 15:54  phsultan       Note Added: 0071477                          
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