[asterisk-bugs] [Asterisk 0005413]: [patch] 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:                
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
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:             2005-10-09 10:36 CDT
Last Modified:              2008-10-18 08:11 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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 (0093938) malaiwah (reporter) - 2008-10-18 08:11
 http://bugs.digium.com/view.php?id=5413#c93938 
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Tested using http://svn.digium.com/svn/asterisk/team/jpeeler/srtp/ (most
up-to-date branch for srtp that I found). I builded it on Ubuntu 8.04 LTS
32-bits.

root at buildbox:/usr/src/asterisk-srtp# dpkg -l | grep libsrtp
ii  libsrtp1-dev                      1.4.4~dfsg-1                Secure
RTP (SRTP) and UST Reference Implemen

I did not try the res_mikey component as I do not think Grandstream
devices are using it.
I had to include "srtp/srtp_priv.h" header file in "res/res_srtp.c" to
make it compile properly (was missing symbols when compiling).
The "ast_srtp_depend" patch was not needed to compile properly.

I'm using the demo extensions.conf file from the samples, I only added a
friend ip sip.conf and changed externip parameters because this testing
server is on a private nat.

Using a GXP2000 firmware 1.1.6.16, SRTP forced in the device
configuration, codec PCMU. All I get is garbage sound.
Using the same GXP device, SRTP disabled, codec PCMU worked as it should.
I will attach some captures that I made during testing (asterisk console
at core verbose 99, gzipped tcpdump).

I tried forcing the GSM codec, but now there is only silence when forcing
SRTP on the Grandstream device. Reverting back to disabling SRTP makes it
work correctly.

I guess the garbled sound could be a side-effect of this warning.

[Oct 18 13:08:36] WARNING[23522]: sdp_crypto.c:161 sdp_crypto_activate:
Could not set remote SRTP policy

But I don't know how to fix it. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-10-18 08:11 malaiwah       Note Added: 0093938                          
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