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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=5413 
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Reported By:                mikma
Assigned To:                twilson
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Target Version:             1.6.3.0
Asterisk Version:           SVN 
Regression:                 No 
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:              2009-05-05 04:31 CDT
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Summary:                    [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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 (0104192) aanderson (reporter) - 2009-05-05 04:31
 http://bugs.digium.com/view.php?id=5413#c104192 
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i'm using the branch for about three months now, ***without using srtp***
(no supported phones yet), but we've done some 3000 calls without any
problems, so the impact on the rest of asterisk seems minimal. What has to
be done to finally (after allmost five years) get this into trunk and maybe
1.6.3? Are there any showstoppers, or can some people have a look at this
in the reviewboard...? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-05 04:31 aanderson      Note Added: 0104192                          
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