[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 3 11:23:01 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=5413
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Reported By: mikma
Assigned To:
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Project: Asterisk
Issue ID: 5413
Category: Channels/chan_sip/NewFeature
Reproducibility: N/A
Severity: feature
Priority: normal
Status: confirmed
Target Version: 1.8
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 48491
Request Review:
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Date Submitted: 2005-10-09 10:36 CDT
Last Modified: 2010-03-03 11:22 CST
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Summary: [patch] [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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(0118866) twilson (administrator) - 2010-03-03 11:22
https://issues.asterisk.org/view.php?id=5413#c118866
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actually, looking more closely at the debug output, you aren't making an
outbound call, just calling in and listening to a sound file. This has
always worked for me even without any fixes, so I'm a little confused as to
why it hasn't been working for you.
The only difference is that I'm not using TLS on my grandstream (GXP2000
the firmware currently on it doesn't support it), but that shouldn't be
effecting the encrypted audio.
Issue History
Date Modified Username Field Change
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2010-03-03 11:22 twilson Note Added: 0118866
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