[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=5413 
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Reported By:                mikma
Assigned To:                
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Target Version:             1.8
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Request Review:              
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Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2010-03-22 23:59 CDT
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Summary:                    [patch] [branch] Secure RTP (SRTP)
Description: 
This patch adds initial support for secure RTP using libsrt[1]. It can
be used in for example an implementation of the sdecriptions draft[2].

[1] http://srtp.sourceforge.net/srtp.html
[2]
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions-12.txt


Update (16/03/2010): Branch against trunk is located here
http://svn.asterisk.org/svn/asterisk/team/group/srtp_reboot

*** IF TESTING, PLEASE USE THE ABOVE BRANCH AND NOT THE PATCHED ATTACHED
TO THIS ISSUE AS THEY ARE OUT OF DATE ***
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Relationships       ID      Summary
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related to          0010129 Module SRTP can't loaded
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 (0119692) hemanshurpatel (reporter) - 2010-03-22 23:59
 https://issues.asterisk.org/view.php?id=5413#c119692 
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hi twilson
What you are doing is using the same a_crypto tag two times for audio and
video. I have already tried this and even created a complete new v_crypto
tag.
in both cases result was same as above.

in channel/chan_sip.c line no 25086 in function call
sdp_crypto_process(p->srtp->crypto, a, p->rtp)

Now this function will set the crypto for audio rtp session only and so
the ssrc value which is used to identify the encrypted session.
But when sip_rtp_read function is called, it passes p->rtp and p->vrtp for
audio and video. now as we setup crypto for p->rtp only in your case, video
packets wont work.i havent test this new srtp_reboot, just was going
through code. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-22 23:59 hemanshurpatel Note Added: 0119692                          
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