[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]

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:                     new
Asterisk Version:           1.8.4 
JIRA:                        
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:              2011-05-12 03:57 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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 (0134821) schmidts (manager) - 2011-05-12 03:57
 https://issues.asterisk.org/view.php?id=19281#c134821 
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my patch would solve another problem which is in this invite. the invite
offers two simultan audio streams, one with normal RTP the other with SRTP
but the second offer doesnt have a port number but a=sendrecv.

BTW asterisk couldnt support two streams of the same type like audio, only
different streams like audio + video.

the RFC definition says this offer is valid only the port is not right,
but normally asterisk should handle this right. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-05-12 03:57 schmidts       Note Added: 0134821                          
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