[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 8 07:14:38 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: 2010-03-08 07:14 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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(0119085) hemanshurpatel (reporter) - 2010-03-08 07:14
https://issues.asterisk.org/view.php?id=5413#c119085
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as far as handling SRTP data by asterisk is concerned i could make only
audio calls....
it gives 488 in case when i enable video.
With video it fails.but it works nice with audio.
can anyone help in finding the solution with video.
i had also implement the same logic with asterisk 1.4.0, in that not even
audio works if i ask asterisk to handle to data. Proxying thing work great.
Issue History
Date Modified Username Field Change
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2010-03-08 07:14 hemanshurpatel Note Added: 0119085
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