[Asterisk-bugs] [Asterisk 0005413]: [patch] Secure RTP (SRTP)
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Jul 20 23:11:22 CDT 2007
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: oej
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Project: Asterisk
Issue ID: 5413
Category: Core/RTP
Reproducibility: N/A
Severity: feature
Priority: normal
Status: ready for testing
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 48491
Disclaimer on File?: Yes
Request Review:
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Date Submitted: 10-09-2005 10:36 CDT
Last Modified: 07-20-2007 23:11 CDT
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Summary: [patch] 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
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Relationships ID Summary
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related to 0010129 Module SRTP can't loaded
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kumarvivek_24 - 07-20-07 23:11
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hello all,
i have successfully add srtp patch with my asterisk tar version . But i
want to do some more.
Is it possible that in case of srtp call between two snom phone, rtp will
pass direct peer to peer,without asterisk is in the path.
rtp
snom <====> snom
please help me.
thanks to all.
Issue History
Date Modified Username Field Change
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07-20-07 23:11 kumarvivek_24 Note Added: 0067689
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