[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:                otherwiseguy
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Target Version:             1.6.3
Asterisk Version:           SVN 
Regression:                 No 
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:              2009-01-09 10:01 CST
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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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 (0097307) phsultan (manager) - 2009-01-09 10:01
 http://bugs.digium.com/view.php?id=5413#c97307 
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> pshultan: Asterisk should be encrypting those calls if there is an SAVP
profile. 
> It sounds like they are using the "Send RTP/AVP but add a crypto line"
method
> (which is totally not valid according to the specs, but I digress). I
meant to 
> remove asterisk *sending* calls that way, but didn't mean to remove
accepting 
> calls that way. I will try to get it to go back to encrypting calls with
an 
> optional SRTP specified this way.

This is indeed the method they chose, at least on the phones (+ versions)
I tested. In my case, ignoring the 'secure_audio' parameter (set if
RTP/SAVP is received) solved the problem. So I commented out this section
:
if (!secure_audio && p->srtp) {
  ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without
it!\n");
  return -2;
}

That's a quick and dirty hack, maybe a test to see if the call is outgoing
would be more relevant :
if (!secure_audio && p->srtp && p->outgoing_call == TRUE) 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-09 10:01 phsultan       Note Added: 0097307                          
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