[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:                otherwiseguy
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Target Version:             1.6.3
Asterisk Version:           SVN 
Regression:                 No 
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:              2009-01-11 17:20 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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 (0097455) otherwiseguy (administrator) - 2009-01-11 17:20
 http://bugs.digium.com/view.php?id=5413#c97455 
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notthematrix:
Not everyone who has phones that use tls or support encryption will want
to use it on any/all calls. Enabling encryption will have a fairly
significant performance impact, so the idea is to let people decide for
themselves what calls to allow it on.  It isn't Asterisk's job to determine
what people want to do.  That is why all of the stuff with the dialplan
variables, etc. were implemented (at least I assume, since I didn't
originally write any of it).  I don't think we will *ever* make encryption
automatic on an outbound call.  It will be determined via the dialplan and
enabled by the user based on the information that is made available to the
dialplan.

Is there a particular use case that you are trying to implement that isn't
currently capable?  I'm having trouble understanding what it is that you
are wanting done.  But, I'm a little slow sometimes.  :-) 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-11 17:20 otherwiseguy   Note Added: 0097455                          
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