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

[noreply at bugs.digium.com]

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:                jpeeler
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Core/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
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:             10-09-2005 10:36 CDT
Last Modified:              04-13-2008 09:34 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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---------------------------------------------------------------------- 
 rjain - 04-13-08 09:34  
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All, I tested the following branch:
http://svn.digium.com/svn/asterisk/team/jpeeler/srtp/ 

against SNOM 360 as well as Grandstream phones using SDES for SRTP key
exchange. I'm not hearing any voice. The SIP/SDP signaling looks fine and I
see SRTP packets flowing back and forth. I'm doing a simple 1-party call
with Asterisk doing a playback.

What could I be missing? What kind of a debug/trace information would be
helpful? Help? 

Issue History 
Date Modified   Username       Field                    Change               
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04-13-08 09:34  rjain          Note Added: 0085422                          
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