[asterisk-bugs] [Asterisk 0005413]: [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Dec 22 07:08:32 CST 2008
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 07:08 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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(0096779) notthematrix (reporter) - 2008-12-22 07:08
http://bugs.digium.com/view.php?id=5413#c96779
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I did play a bit with it yesterday.
I have a grandstream HT-503 and I Foreced SRTP so it would only work if
SRTP worked at asterisk this in combination with TLS.
And could call out.
do echo test.
but when I tryed to call It crached with a seg fault.
but Iam not sure it had to do with the srtp code because it also crahed
whitout srtp active.
a workble pachh make for 1.6.1-beta3 or beta4 would be a nice because
Beta3 works corectly with tls enabled.
so I could test that with my grandstream ht-503 firmware 10.0.15
Issue History
Date Modified Username Field Change
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2008-12-22 07:08 notthematrix Note Added: 0096779
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