[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:                
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2008-12-17 17:12 CST
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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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 (0096597) otherwiseguy (administrator) - 2008-12-17 17:12
 http://bugs.digium.com/view.php?id=5413#c96597 
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Ok, there were some calls to sendto() in main/rtp.c that needed to be
changed to the helper function rtp_sendto() which actually does the check
for whether or not asterisk should try to encrypt outbound audio.  Before
the change, asterisk would receive the encrypted audio but not encrypt the
outgoing stream.  On my polycom, anyay, this resulted in no audio. 
Revision 165396 has the fix.

Anyone else still hanging on to test this stuff? 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-17 17:12 otherwiseguy   Note Added: 0096597                          
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