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

noreply at bugs.digium.com noreply at bugs.digium.com
Fri Jul 20 23:11:22 CDT 2007


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:                oej
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Core/RTP
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): trunk 
SVN Revision (number only!): 48491 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             10-09-2005 10:36 CDT
Last Modified:              07-20-2007 23:11 CDT
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Summary:                    [patch] 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

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Relationships       ID      Summary
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related to          0010129 Module SRTP can't loaded
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---------------------------------------------------------------------- 
 kumarvivek_24 - 07-20-07 23:11  
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hello all,

i have successfully add srtp patch with my asterisk tar version . But i
want to do some more. 
Is it possible that in case of srtp call between two snom phone, rtp will
pass direct peer to peer,without asterisk is in the path.
              
               rtp
        snom <====> snom 

please help me.
thanks to all. 

Issue History 
Date Modified   Username       Field                    Change               
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07-20-07 23:11  kumarvivek_24  Note Added: 0067689                          
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