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

[Asterisk Bug Tracker]

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:                twilson
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Target Version:             1.6.5.0
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 48491 
Request Review:              
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Date Submitted:             2005-10-09 10:36 CDT
Last Modified:              2009-04-14 11:31 CDT
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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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 (0103223) uc25393 (reporter) - 2009-04-14 11:31
 http://bugs.digium.com/view.php?id=5413#c103223 
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Hello all, I'm trying to use SRTP in Asterisk with a Twinkle phone. I've
done all the steps given in
http://www.voip-info.org/wiki/view/Asterisk+SRTP. I've built the SRTP
library and then I've get asterisk from
https://svn.digium.com/svn/asterisk/team/group/srtp and I've built it. Then
I've modified my sip.conf and my extension.conf in this way

sip.conf

[9000]
type=friend
qualify=yes
secret=guessthis
nat=no
host=dynamic
canreinvite=no
context=phones
srtpcapable=yes

extension.conf

exten => 9000,1,Set(_SIP_SRTP_SDES=1)
exten => 9000,n,Set(_SIPSRTP=enable)
exten => 9000,n,Set(_SIPSRTP_CRYPTO=enable)
exten => 9000,n,Dial(SIP/9000,30,Tt)
exten => 9000,n,Hangup()

I don't know how to view if the call is encrypted or no. Asterisk don't
return any message about if the call is encrypted or no.I've used the
wireshark to view the frames but I can't view nothing about crypto.

Thank you in advance 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-14 11:31 uc25393        Note Added: 0103223                          
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