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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Jun 15 09:18:31 CDT 2009


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:                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:              2009-06-15 09:18 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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---------------------------------------------------------------------- 
 (0106398) twilson (administrator) - 2009-06-15 09:18
 https://issues.asterisk.org/view.php?id=5413#c106398 
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I'm not sure it is dead...I just don't have very much time to be able to
work on it now.  And the rtp-engine changes means there is a bit of work to
do.  This is one of those times when I really hope some community member
somewhere can pick up the slack and come up with a patch that merges this
up to trunk--because other than that, I'd say it is pretty much ready to
go.

For an example of where to begin with the changes, look at the original
rtp_engine commit and look at the changes to channel drivers and
res_rtp_asterisk.c.  To get the patch, svn diff -c186078
http://svn.digium.com/svn/asterisk/trunk 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-06-15 09:18 twilson        Note Added: 0106398                          
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