[asterisk-bugs] [Asterisk 0005413]: [branch] Secure RTP (SRTP)

[Asterisk Bug Tracker]

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:                twilson
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Target Version:             1.6.3.0
Asterisk Version:           SVN 
Regression:                 No 
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:              2009-06-26 03:46 CDT
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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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 (0107017) aanderson (reporter) - 2009-06-26 03:46
 https://issues.asterisk.org/view.php?id=5413#c107017 
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Maybe someone who monitors this bug has missed the posting on asterisk-dev

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The time to get the SRTP branch merged to trunk is upon us! One of the
oldest
issues in the Asterisk issue tracker (issue number 5413,
https://issues.asterisk.org/view.php?id=5413) is nearly ready to get
merged to
trunk, which means it could find itself in the next major release of
Asterisk,
most likely Asterisk 1.6.3.0. However, some work is still required before
we
can get it merged.

Terry Wilson has, until recently, been working on the issue getting it
ready to
be committed, and has performed all the updates required from the
reviewboard
review (https://reviewboard.asterisk.org/r/191/) and the current code for
SRTP
is sitting in a group branch
(http://svn.digium.com/svn/asterisk/team/group/srtp/).

Since Terry has been pulled away from SRTP to work on a couple of other
larger
projects, he has not been able to update the SRTP branch to be in sync
with
trunk after the rtp-engine commits. This is where you come in!

We're asking for help from the community to provide a patch that will get
the
SRTP work brought back up to date so that it can get merged. A note from
Terry
gives some information on how to get started, "For an example of where to
begin
with the changes, look at the original rtp_engine commit and look at the
changes
to channel drivers and res_rtp_asterisk.c. To get the patch,
svn diff -c186078 http://svn.digium.com/svn/asterisk/trunk".

Once you've got the patch ready for review, please attach it to the
original
issue on the Asterisk issue tracker at
https://issues.asterisk.org/view.php?id=5413

Thanks!
Leif Madsen
Asterisk issue marshal

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Issue History 
Date Modified    Username       Field                    Change               
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2009-06-26 03:46 aanderson      Note Added: 0107017                          
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