[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:              2008-12-18 15:02 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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 (0096661) umberto71 (reporter) - 2008-12-18 15:02
 http://bugs.digium.com/view.php?id=5413#c96661 
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I've done a working patch for asterisk 1.6.0.1. release
I've tested with Snom, Astra Grandstream real phones and snom softphone
and phonerlite. Audio works in full duplex, music on hold, transfer,
conference work fine. (only with _SIPSRTP_CRYPTO )
I've merged Jpeeler trunk. I don't know if I can post this patch here,
please send me some info . 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-18 15:02 umberto71      Note Added: 0096661                          
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