[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:              2010-02-21 15:23 CST
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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 (17/12/2008): Branch against trunk is located here
http://svn.digium.com/svn/asterisk/team/group/srtp
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Relationships       ID      Summary
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related to          0010129 Module SRTP can't loaded
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 (0118319) twilson (administrator) - 2010-02-21 15:23
 https://issues.asterisk.org/view.php?id=5413#c118319 
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Everything is still moving around a lot with the code. The dialplan
functions are all different, there are no SIPSRTP channel variables, etc.
So things aren't quite expected to work right now. I hope to have things
settled down in the next couple of weeks. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-21 15:23 twilson        Note Added: 0118319                          
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