[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Feb 21 07:27:08 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: 2010-02-21 07:26 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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(0118314) notthematrix (reporter) - 2010-02-21 07:26
https://issues.asterisk.org/view.php?id=5413#c118314
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Hi twilon
tanx for the info we are very happy the project is alive again... :)
but iam also want to tell that we found an issue with the grandstream
ht-50x devices.
In the old version http://svn.digium.com/svn/asterisk/team/group/srtp
grandstream and forced SRTP worked very well.
but when we tried the new version you get white noise all the time..
Wat has changed?
We are willing to test etc see the diffrence , we also have some good
contacts with the grandstream dev team.
So it must be possible to fix this and let the ht-50x SRTP work well with
asterisk 1.8
If neded we have equipent like a mannaged switch etc.
hope you can fix this problem.
Issue History
Date Modified Username Field Change
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2010-02-21 07:26 notthematrix Note Added: 0118314
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