[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
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:              2008-12-22 23:01 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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 (0096875) otherwiseguy (administrator) - 2008-12-22 23:01
 http://bugs.digium.com/view.php?id=5413#c96875 
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notthematrix: the SIP_SRTP_SDES variable doesn't actually exist anymore. 
Currently, it looks like that for the caller's leg of the call asterisk
will encrypt that leg of the call no matter what if it offers encryption. 
The variables currently only control what happens w/ the outgoing leg of
the call.  So if you have two phones and call from one to the other, if the
caller supports SRTP it will be encrypted.  If you have
SIPSRTP_CRYPTO=disable set, then the leg between asterisk and the callee
will not be encrypted.  Currently there isn't anything available which
would allow the call to be rejected if both legs are not capable of
encryption.

There is currently a bug where even if the dialplan specifies optional,
Asterisk will send a a=crypto line, but in a normal RTP/AVP (instead of
RTP/SAVP) offer which makes my polycom reject the call if it is set not to
accept SRTP calls.  Fixing that is easy enough, but I agree a lot of the
dialplan variable / setting stuff needs to be reworked.  I'll try to get it
all hammered out into something more useable tomorrow. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-22 23:01 otherwiseguy   Note Added: 0096875                          
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