[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:              06-25-2008 02:08 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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---------------------------------------------------------------------- 
 kactus - 06-25-08 02:08  
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Hello JSmith

Please find the SipDebugCleaned.txt attached

This is the wosrt case of it talking to the cisco

Psip - SIP -> A1.6 – SIP -> PRIcisco

Let me know if you need any thing else, the 600+ lines of 
[Jun 25 14:16:40] DEBUG[76060]: channel.c:1129 channel_find_locked:
Avoiding initial deadlock for channel '0x8800000' 
might have something to do with it if I was a betting man.

Cheers 

Issue History 
Date Modified   Username       Field                    Change               
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06-25-08 02:08  kactus         Note Added: 0089200                          
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