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

[Asterisk Bug Tracker]

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:                murf
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Project:                    Asterisk
Issue ID:                   12708
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
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:             2008-05-22 20:26 CDT
Last Modified:              2008-08-07 13:05 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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---------------------------------------------------------------------- 
 (0091208) putnopvut (administrator) - 2008-08-07 13:05
 http://bugs.digium.com/view.php?id=12708#c91208 
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Repost of what I said in response to PJ on the asterisk-users list:

"If you check ast_answer in channel.c of trunk, you can see that it calls
__ast_answer(chan, 500). The 500 there is a 500 ms delay that occurs before
calling the channel's answer callback. In the case of SIP, this would
indeed mean that there is a 500 ms delay between receiving the 200 OK from
the callee and sending a 200 OK to the caller."

The initial commit that added this delay was 50538, and it was modified
slightly with commit 50571.

Then issue http://bugs.digium.com/view.php?id=12924 came up and the delay
behavior was changed again with
commits 127113 and 127157.

The reason for adding the delay in the first place was to allow time for
the media to be set up after the call had been answered. To be honest, with
where the delay is now, it really doesn't make a lot of sense for it to be
there since it's not actually allowing for media to be set up (since
Asterisk hasn't even sent a 200 OK to the caller yet). 

Issue History 
Date Modified    Username       Field                    Change               
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2008-08-07 13:05 putnopvut      Note Added: 0091208                          
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