[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 03:37 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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---------------------------------------------------------------------- 
 (0134819) schmidts (manager) - 2011-05-12 03:37
 https://issues.asterisk.org/view.php?id=19281#c134819 
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btw pcap isnt recommended for issues so i copy the invite in here:

v=0
o=MxSIP 0 1849763221 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 29100 RTP/AVP 8 0 18 4 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
a=sendrecv
m=audio 0 RTP/SAVP 8 0 18 4 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
a=sendrecv 

Issue History 
Date Modified    Username       Field                    Change               
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2011-05-12 03:37 schmidts       Note Added: 0134819                          
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