[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:             1.6.5.0
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:              2009-04-21 03:50 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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 (0103516) kla960 (reporter) - 2009-04-21 03:50
 http://bugs.digium.com/view.php?id=5413#c103516 
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@vrban the PhonerLite dont write "SAVP" in the m=audio 5062 RTP/AVP 8 0 2 3
97 110 111 9 101 line in the sdp. Thats why you get the SIP 488 error

that's of course the srtp is set allways optional in phonerlite. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-21 03:50 kla960         Note Added: 0103516                          
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