[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:              2009-01-09 09:39 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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 (0097306) otherwiseguy (administrator) - 2009-01-09 09:39
 http://bugs.digium.com/view.php?id=5413#c97306 
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pshultan: Asterisk should be encrypting those calls if there is an SAVP
profile. It sounds like they are using the "Send RTP/AVP but add a crypto
line" method (which is totally not valid according to the specs, but I
digress).  I meant to remove asterisk *sending* calls that way, but didn't
mean to remove accepting calls that way.  I will try to get it to go back
to encrypting calls with an optional SRTP specified this way.

notthematrix: I'm not maintaining a useragent db.  Of course, all the
information is available to the user to be able to maintain there own if
they want, I just think that there are easier ways for most circumstances. 
My job is to make enough information available to the user to let them do
whatever they want.  :-)  That is what the srtpcapable flag is for in
sip.conf, the administrator knows which phones support encryption and need
it enabled at least on some calls.  And on incoming calls, if their is an
encryption offer, we will automatically encrypt.  It just sounds like I may
have broken that for phones that don't adhere to the RFCs w/ regard to
SDP/SRTP. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-09 09:39 otherwiseguy   Note Added: 0097306                          
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