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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=5413 
====================================================================== 
Reported By:                mikma
Assigned To:                otherwiseguy
====================================================================== 
Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Target Version:             1.6.3
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Disclaimer on File?:        Yes 
Request Review:              
====================================================================== 
Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2009-01-11 18:01 CST
====================================================================== 
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
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0010129 Module SRTP can't loaded
====================================================================== 

---------------------------------------------------------------------- 
 (0097456) notthematrix (reporter) - 2009-01-11 18:01
 http://bugs.digium.com/view.php?id=5413#c97456 
---------------------------------------------------------------------- 
I understand the preformance inpact problem.
But what i suggest is to default to rtpcapable=yes
when the device uses TLS to register , since 99% wants to use SRTP when
TLS is used annyway.
for the ones that dont they can always set srtpcapable=no
the same options in the dailplan offcource but when default when a device
registers with TLS is set to rtpcapable=YES
It wil save space in configuration files.
since only the ones who dont want srtp need to add srtpcapable=no
an other option is to make tho option usesrtpwhentls= yes | no
this could then be placed in the [general] settings of sip.conf for a
general setting for all devices.
thinking about this the most liberal option would be "Use srtp when tls
yes / no"
usesrtpwhentls= yes | no  
wil solve all needs in one setting for most poeple.
of this can be put in the [general] section of sip.conf most poeple would
be happy. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-11 18:01 notthematrix   Note Added: 0097456                          
======================================================================