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

noreply at bugs.digium.com noreply at bugs.digium.com
Sun Feb 17 23:50:20 CST 2008


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:              02-17-2008 23:49 CST
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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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---------------------------------------------------------------------- 
 Xianglin - 02-17-08 23:49  
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Hi there,
   I am doing a research project on SRTP, with Asterisk and Minisip
softphone. I followed the installation instruction at
http://www.voip-info.org/wiki/view/Asterisk+SRTP

however, when I make phone calls, SRTP seems not working, as the media is
still carried out by RTP. 

Could anyone please let me know how to "enabloe" SRTP? 
I will  really appreciate your help. 
Thank you very much. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-17-08 23:49  Xianglin       Note Added: 0082455                          
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