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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/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
Target Version:             1.8
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
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:              2010-03-05 00:30 CST
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Summary:                    [patch] [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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 (0119014) hemanshurpatel (reporter) - 2010-03-05 00:30
 https://issues.asterisk.org/view.php?id=5413#c119014 
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hi, i had started following SRTP via
http://www.voip-info.org/wiki/view/Asterisk+SRTP page.

I am using grandstream gvx3140 with latest firmware 1.0.3.X i guess.
Initially they were not making calls, i mean asterisk was not allowing
call, as GS was sending xxxxxxx|2^32 as last thing in crypto value...where
asterisk was saying lifetime support is not allowed.....

Then i commented that part, and applied few more hacks, and make asterisk
doing SRTP Proxy work. Which initially wasnt allowed as SIPSRTP is set.
So as on now with GS gxv3140 SRTP is working nice.
got lot of info on this page itself.

But have few doubts...as far as proxying is concerned SRTP is working. but
is i want my media to pass through asterisk as in case of LI, asterisk is
not making calls.

Problem of lifetime support is an issue, and if i comment that portion and
go ahead then asterisk forward new crypt value without |2^32 at end to
other device... and Grandstream at other end send 48X something saying pls
chose diff vocoder....

How can i make astersik handle SRTP as well....with |2^32 stuff in end....
coz i need this as well for LI.
Any inputs will be appreciated.

Cheers
Hemanshu 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-05 00:30 hemanshurpatel Note Added: 0119014                          
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