[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:                oej
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Project:                    Asterisk
Issue ID:                   5413
Category:                   Core/RTP
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): trunk 
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:              12-12-2007 04:18 CST
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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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---------------------------------------------------------------------- 
 oarpvfpre - 12-12-07 04:18  
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Hi there,

I have successfully compiled SVN-trunk-r81432M with
ast_srtp_r81432_mikey_r3412.patch and connected two minisip. A secure call
to the number 01 or the echo test works nice. When I try to make a secure
call between the two phones, Asterisk show a message "Cannot native bridge
in SRTP".

I have try to set "canreinvite=no" in sip.conf, but it still doesn't work,
what is the next step to enable secure calls between two phones?

--------------------------------------
configuration in sip.conf 
--------------------------------------
[isec_01]
type=friend
username=isec_01
secret=isec
host=dynamic
context=tutorial
canreinvite=no
nat=no

--------------------------------------
configuration in extensions.conf 
--------------------------------------
[tutorial]
exten => 01, 1, Set(_SIP_SRTP_SDES=1)
exten => 01, n, Set(_SIPSRTP=optional)
exten => 01, n, Set(_SIPSRTP_CRYPTO=enable)
exten => 01, n, Dial(SIP/isec_01)

--------------------------------------
message in Asterisk CLI
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[Dec 12 17:50:04] NOTICE[2562]: sdp_mikey.c:112 sdp_mikey_setup: Using
MIKEY PSK isec
    -- Executing [01 at tutorial:1] Set("SIP/isec_02-08374fc0",
"_SIP_SRTP_SDES=1") in new stack
    -- Executing [01 at tutorial:2] Set("SIP/isec_02-08374fc0",
"_SIPSRTP=optional") in new stack
    -- Executing [01 at tutorial:3] Set("SIP/isec_02-08374fc0",
"_SIPSRTP_CRYPTO=enable") in new stack
    -- Executing [01 at tutorial:4] Dial("SIP/isec_02-08374fc0",
"SIP/isec_01") in new stack
  == Using TOS bits 0
  == Using CoS mark 5
[Dec 12 17:50:04] NOTICE[2562]: chan_sip.c:3554 sip_call: SIPSRTP_CRYPTO
[Dec 12 17:50:04] NOTICE[2562]: chan_sip.c:3541 sip_call: SIPSRTP
    -- Called isec_01
    -- SIP/isec_01-0837e730 is ringing
--- set_address_from_contact host '140.125.84.30'
    -- SIP/isec_01-0837e730 answered SIP/isec_02-08374fc0
[Dec 12 17:50:08] NOTICE[2562]: rtp.c:3924 ast_rtp_bridge: Cannot native
bridge in SRTP. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-12-07 04:18  oarpvfpre      Note Added: 0075255                          
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