[asterisk-bugs] [Asterisk 0014707]: No audio from Gtalk to Asterisk

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14707 
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Reported By:                tootai
Assigned To:                phsultan
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Project:                    Asterisk
Issue ID:                   14707
Category:                   Channels/chan_gtalk
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.24 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-03-20 05:28 CDT
Last Modified:              2009-05-19 08:27 CDT
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Summary:                    No audio from Gtalk to Asterisk
Description: 
When calling from Gtalk client to an Asterisk user we have no audio in both
direction. We have calling signal in GTalk client until called party
answers, on callee side phone is ringing normally and that's all. In logs
we have:

[Mar 20 10:20:08] VERBOSE[20006] logger.c:     --
Local/821 at ServiceNumbers-754b,1 answered Gtalk/ivan.trudnai-2df4
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '-1'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '20'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '20'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '-1'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '20'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '20'
[Mar 20 10:20:49] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '20'
[Mar 20 10:20:49] VERBOSE[20007] logger.c:     -- Executing
[h at ServiceNumbers:1] GotoIf("Local/821 at ServiceNumbers-754b,2", "0?end") in
new stack
[Mar 20 10:20:49] VERBOSE[20007] logger.c:     -- Executing
[h at ServiceNumbers:2] Hangup("Local/821 at ServiceNumbers-754b,2", "") in new
stack
[Mar 20 10:20:49] VERBOSE[20007] logger.c:   == Spawn h extension
(ServiceNumbers, h, 2) exited non-zero on
'Local/821 at ServiceNumbers-754b,2'
[Mar 20 10:20:49] VERBOSE[20007] logger.c:   == Spawn extension
(ServiceNumbers, 104, 7) exited non-zero on
'Local/821 at ServiceNumbers-754b,2'
[Mar 20 10:20:49] NOTICE[20006] chan_gtalk.c: Don't know how to indicate
condition '20'
[Mar 20 10:20:49] VERBOSE[20006] logger.c:   == Spawn extension
(from-GTALK, tootainet at gmail.com, 4) exited non-zero on
'Gtalk/ivan.trudnai-2df4'
[Mar 20 10:20:50] NOTICE[3937] chan_gtalk.c: Whoa, didn't find call!

The last NOTICE seems strange.

-- 
Daniel
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---------------------------------------------------------------------- 
 (0104996) peterx86 (reporter) - 2009-05-19 08:27
 https://issues.asterisk.org/view.php?id=14707#c104996 
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Hi This is Peter. For my hobby, I tried myself to write some code to call
from gmail to sip.

(1)what i do is to test my concept, i have a separate process to work as
back to back call gateway. The gmail call to my program (which works as
xmpp external component), my program call asterisk through SIP and do
session negotiate. 

(2)Ring and answer all works. I am using pcmu because i know google talk
use different payload type constants except a few types. Anyway, for
testing concept, it is fine.

(3)After answer, only SIP phone can hear voice. It is within my
expectation because i have not do SIP re-invite to try different
candidates. Even i do re-invite, it can only support pcmu,a types. It is
not what i want.

Now, i stopped. I am thinking what is the simplest way to do? google
jingle has very complex p2p negotiation. Also, i must do relay to do
payload rewriting. 

Question:
(A)Must i simulate the p2p negotiation process? 
(B)Or just relay.

Because i am not familiar with asterisk channel development, i have no
idea how to go through the chan_gtalk.c file.

By the way, if i want to test chan_gtalk, which asterisk version should i
use?

Can you share some idea?

Best Regards
Peter 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-19 08:27 peterx86       Note Added: 0104996                          
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