[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:              07-03-2008 04:33 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 - 07-03-08 04:33  
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Hello Ranahimal 

Yes we are using realtime peers through odbc to a MSSQL database with
freetds 0.64 as our driver.

Unfortunately some of the things we are doing do not allow us to run
rtcachefriends=yes as we need to be able to programatically manage end
users sip connections (such as password and context) on the fly without
requiring a reload.

Also our sip service interconnect provider does not take registration,
just accepts connections from an ip address so even if we wanted to we
couldn't cache the connection. (they are also our ISP so this method is
secure)

I didn't notice until you pointed it out that it appears to be querying
the database several times after it begins to dial, but this does not
explain why when it talks through the 1.2 box there is much less delay (as
this connection is not cached either) unless SIP and IAX code differences
are causing this to happen. 

What would you like to to try for you, to help you diagnose the cause of
this?

Thanks you for your time in investigating this issue. 

Issue History 
Date Modified   Username       Field                    Change               
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07-03-08 04:33  kactus         Note Added: 0089660                          
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