[asterisk-bugs] [Asterisk 0014126]: Using dtmfmode=info AND canreinvite=yes (both in sip.conf) AND dynamic features (features.conf/Dial() with w or W flags)

Mon Dec 22 16:00:30 CST 2008

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14126 
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Reported By:                malaiwah
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14126
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.23-rc3 
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:             2008-12-22 15:22 CST
Last Modified:              2008-12-22 16:00 CST
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Summary:                    Using dtmfmode=info AND canreinvite=yes (both in
sip.conf) AND dynamic features (features.conf/Dial() with w or W flags)
Description: 
If Asterisk is configured for SIP INFO and re-invites, if one wants to
enable dynamic features on a Dial application call, Asterisk will stay in
the audio path for all the conversation.

This behavior should be correct for in-band DTMF and RFC2833 but for SIP
INFO, it could still send a re-invite and get back the voice path IF a
dynamic feature is engaged (in my case, it is
http://bugs.digium.com/view.php?id=3 for quick recording of the
conversation).
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---------------------------------------------------------------------- 
 (0096849) malaiwah (reporter) - 2008-12-22 16:00
 http://bugs.digium.com/view.php?id=14126#c96849 
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Hi Russell.

I was giving numbers by heart, but there are various evidences in the
forums of conccurent call limits like this, but mainly this is what we find
on voip-info.org (you might edit the page if this info is bogus,
http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning):

{{{
The only rule of thumb we appear to be able to provide is this: Asterisk
1.2 start to run into problems around 220 concurrent SIP calls. Asterisk
1.4 scales much better and can handle nearly double the call setups/second
as well as total concurrent traffic. Moreover, Early testing of Asterisk
1.6 using hash tables shows a SIP performance increase compared to 1.4 of
factor 3 to 4. 
}}}

So 1.2=200 bridged calls, 1.4=400 bridged calls and 1.6=1200 bridged
calls?

I might give a try on Asterisk 1.6 then, but anyway my case stand still
Asterisk could re-invite audio paths when peer is using dtmfmode=info.

I will create a simple test case and post it back then. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-22 16:00 malaiwah       Note Added: 0096849                          
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