[asterisk-bugs] [Asterisk 0005413]: [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Jan 9 11:55:52 CST 2009
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 11:55 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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(0097326) otherwiseguy (administrator) - 2009-01-09 11:55
http://bugs.digium.com/view.php?id=5413#c97326
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notthematrix: also, we don't encrypt all calls just because the phone
supports it because there is no standard way to offer "optional"
encryption. We will encrypt the caller's leg if encryption is offered, but
we have to know whether or not to offer or not offer encryption to the
callee because there is no standard way to say "encrypt this if you can"
with SRTP. Different vendors use incompatible ways to do it (and I still
think it is silly...you either want a secure call, or you don't), so we
just send an RTP/SAVP request or an RTP/AVP request for now until that gets
sorted out by everyone.
Issue History
Date Modified Username Field Change
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2009-01-09 11:55 otherwiseguy Note Added: 0097326
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