[asterisk-bugs] [Asterisk 0015385]: SIP clients and "internal_timing" not working when silence suppression enabled

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=15385 
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Reported By:                bryanfe
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15385
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 202749 
Request Review:              
Resolution:                 suspended
Fixed in Version:           
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Date Submitted:             2009-06-23 16:05 CDT
Last Modified:              2010-06-10 15:09 CDT
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Summary:                    SIP clients and "internal_timing" not working when
silence suppression enabled
Description: 
Asterisk is not behaving as expected (based upon what documentation I can
read) when SIP clients have "Silence Suppression" turned on and we are
using the internal timer.

Several apps in Asterisk, including Music on Hold, won't send any  
audio to the SIP client, unless the SIP client itself sends audio to  
Asterisk (which it won't do if Silence Suppression is enabled and the  
caller is quiet).

Silence Suppression is important for our application because the caller
will be mostly quiet (listening, and occasionally entering touchtones), and
we can conserve a lot of bandwidth and support more users if we enable
Silence Suppression on the ATA.

My research indicates that if Asterisk is configured to use an internal
timer (such as provided by dahdi_dummy), then this problem should go away,
and SIP clients with Silence Suppression will function correctly.

However, we think we've done everything right in terms of setup, and it
still isn't working.

Here are the relevant data points:

- Asterisk version is 1.6.1 (Revision 202749 from SVN)
- Dahdi (version 2.2.0 rc2) is installed and running. Linux module
"dahdi_dummy" is loaded. No other dahdi modules are loaded.
- Command-line tool "dahdi_test" returns +99% accuracy
- res_timing_dahdi.so is loaded in Asterisk "modules.conf" file
- "internal_timing" is set to "yes" in Asterisk "asterisk.conf" file
- Asterisk CLI command "timing test" works, with reports such as:
- SIP ATA's are Linksys PAP2Ts and Sipura 2102s.

	Attempting to test a timer with 50 ticks per second.
	Using the 'DAHDI' timing module for this test.
	It has been 1019 milliseconds, and we got 51 timer ticks

Based upon my understanding of things, all of the above points to "go"  
with respect to proper support of SIP Silence Suppression in the  
client, but we're just not seeing it.

Symptoms are easily reproducible. When sending audio to the client and
silence suppresion is on, and the client is very quiet, then no audio is
sent. If the client blows into the mic or has any other background or
foreground noise, then Asterisk will continue to send it audio.

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---------------------------------------------------------------------- 
 (0123265) pabelanger (manager) - 2010-06-10 15:09
 https://issues.asterisk.org/view.php?id=15385#c123265 
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Suspended due to lack of activity. Please request a bug marshal in
#asterisk-bugs on the IRC network irc.freenode.net to reopen the issue
should you have the additional information requested.

Further information can be found at
http://www.asterisk.org/developers/bug-guidelines 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-10 15:09 pabelanger     Note Added: 0123265                          
2010-06-10 15:09 pabelanger     Status                   feedback => closed  
2010-06-10 15:09 pabelanger     Resolution               open => suspended   
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