[asterisk-bugs] [Asterisk 0019281]: [patch] Invite with session description that supports both SRTP(SAVP) and RTP(AVP) fails if asterisk is not configured to use SR

Asterisk Bug Tracker noreply at bugs.digium.com
Fri May 13 06:49:57 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19281 
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Reported By:                jacco
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19281
Category:                   Channels/chan_sip/SRTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.4 
JIRA:                       SWP-3458 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-05-12 02:17 CDT
Last Modified:              2011-05-13 06:49 CDT
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Summary:                    [patch] Invite with session description that
supports both SRTP(SAVP) and RTP(AVP) fails if asterisk is not configured to use
SR
Description: 
I was doing an interoperability with a HiPath 3000 V8 M5T SIP
Stack/4.0.26.26 and it failed.

It has to do with an invite with session desription that supports both
SRTP(SAVP) and RTP(AVP); when asterisk is not configured for SRTP it will
not start an RTP session.
Asterisk CLI shows : 
chan_sip.c: Can't provide secure audio requested in SDP offer
and does not continue with setting up rtp

I guess it should provide an insecure audio request en continue with RTP

tested in asterisk 1.8.2.4 and asterisk 1.8.4



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---------------------------------------------------------------------- 
 (0134881) jacco (reporter) - 2011-05-13 06:49
 https://issues.asterisk.org/view.php?id=19281#c134881 
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I've uploaded a patch (issue19281_2.patch) that will also fix
interoperability with a HiPath 3000 V8 M5T SIP Stack/4.0.26.26 and will
make asterisk more compliant with RFC3264.

To be more specific:
it makes astrisk complaint with this line from part 5.1 of RFC3264:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used

This is my first patch (with license) and I realy hope it will make to the
official release 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-13 06:49 jacco          Issue Monitored: jacco                       
2011-05-13 06:49 jacco          Note Added: 0134881                          
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