[asterisk-bugs] [Asterisk 0012708]: Dead air between answer and packet2packet bridge

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12708 
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Reported By:                kactus
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12708
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0-beta8 
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:             05-22-2008 20:26 CDT
Last Modified:              06-24-2008 07:04 CDT
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Summary:                    Dead air between answer and packet2packet bridge
Description: 
Hi we have been testing asterisk 1.6 extensively as we intend to replace
our long in the tooth 1.2 box that acts as our gateway between our offices
and the switched telephony network.

Asterisk 1.6 talks to directly to a cisco call gateway via sip which talks
to the out side world via PRI

One issue that we have noticed repeatedly is that there is a large delay
between when a call is answered and when voice traffic actually flows. The
delay is also asymmetrical and of the scope of about 2 seconds. This is
very noticeable as calling someone generally misses the entire greeting.

Call flow essentially goes like this:
start call -> ringing -> answered (other party start talking “welcome to
company this is Cameron”) -> their voice flows 2 seconds later and we
hear “ameron”

If we talk they can't here anything either at the beginning.

I have been mainly testing this with a snom 190 (have also tried sp962)
connected via sip to the 1.6 box (over nat).

We have also tested this by passing the voice out to one of the larger
voice providers (who also use cisco equipment) and they have stated time
and time again that it is not their end. Both Cisco gateways run
unauthenticated accepting calls from particular ips automatically.

RTP debug information is attached (RTP stats attached to bottom of it.)

Please let me know if you need anything else. We have run this on two 1.6
boxes one running beta 8 the other running beta 9.

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---------------------------------------------------------------------- 
 jsmith - 06-24-08 07:04  
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This issue will be *much* easier to track down if you can provide a SIP
trace of the problem.

Please add "debug" to the "console" line in logger.conf, and then do the
following from the Asterisk CLI:

core set verbose 9
core set debug 4
logger reload
sip set debug

Then go ahead and make your call, and copy/paste the information into a
text file and attach it to this bug report. 

Issue History 
Date Modified   Username       Field                    Change               
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06-24-08 07:04  jsmith         Note Added: 0089141                          
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