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

Asterisk Bug Tracker noreply at bugs.digium.com
Sat Jan 10 06:58:39 CST 2009


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:                otherwiseguy
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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:              
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Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2009-01-10 06:58 CST
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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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---------------------------------------------------------------------- 
 (0097416) notthematrix (reporter) - 2009-01-10 06:58
 http://bugs.digium.com/view.php?id=5413#c97416 
---------------------------------------------------------------------- 
this might help a bit

the discription here is how it is expected to work...

http://wiki.snom.com/Settings/user_savp


    * Description: This setting is effective only when RTP encryption
(SRTP) is also enabled and is used to specify whether the use of the
RTP/SAVP profile by the phone should be off (for backward compatibility),
optional or mandatory.
          o When this setting is set to "mandatory" the phone will offer
and accept only SDPs that contain m= lines with an audio profile of
RTP/SAVP.
          o When this setting is set to "optional", the phone will offer
SDPs containing two m= lines, one with an audio profile of RTP/SAVP the
other with an audio profile of RTP/AVP and it will accept SDPs containing
m= lines with either profile. The RTP/SAVP profile, being the preferred
one, is listed first.
          o Since some SIP proxies cannot handle RTP/SAVP profiles or
multiple m= lines this setting may also be turned off. In this case the
phone will send SDPs containing RTP/AVP audio profiles only. Whether or not
the crypto attribute is included depends on whether RTP encryption is on or
off.
          o Note: When RTP encryption is turned off this setting has no
effect.</p> 

    * Valid values: <off>, <optional>, <mandatory>
    * Default value: off 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-10 06:58 notthematrix   Note Added: 0097416                          
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