[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]

The following issue has been UPDATED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=19281 
====================================================================== 
Reported By:                jacco
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2011-05-12 02:17 CDT
Last Modified:              2011-05-12 07:20 CDT
====================================================================== 
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



====================================================================== 

---------------------------------------------------------------------- 
 (0134830) lmadsen (administrator) - 2011-05-12 07:20
 https://issues.asterisk.org/view.php?id=19281#c134830 
---------------------------------------------------------------------- 
I have added a note from the asterisk-dev mailing list from Kevin Fleming
about the media architecture. I wonder if we can do anything here about
this, because as Kevin mentions, there is no mechanism to "fail over" to a
non-secure stream. I may be that Asterisk is actually RFC compliant in this
case. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-12 07:20 lmadsen        Note Added: 0134830                          
2011-05-12 07:20 lmadsen        Additional Information Updated                  
 
======================================================================