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

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Oct 2 10:07:00 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=5413 
====================================================================== 
Reported By:                mikma
Assigned To:                oej
====================================================================== 
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:              
====================================================================== 
Date Submitted:             10-09-2005 10:36 CDT
Last Modified:              10-02-2007 10:06 CDT
====================================================================== 
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

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0010129 Module SRTP can't loaded
====================================================================== 

---------------------------------------------------------------------- 
 kla960 - 10-02-07 10:06  
---------------------------------------------------------------------- 
I'm trying the latest patch with Fritz!box fon 7170 and Phoner an
Softphone. Both can handle SRTP via sdp_crypto. I always get the following
error. Why???

Executing [1235 at default:1] Set("SIP/1236-0a0c0530", "_SIPSRTP=optional")
in new stack
    -- Executing [1235 at default:2] Set("SIP/1236-0a0c0530",
"_SIPSRTP_CRYPTO=enable") in new stack
    -- Executing [1235 at default:3] Dial("SIP/1236-0a0c0530", "SIP/1235,20")
in new stack
  == Using TOS bits 0
  == Using CoS mark 5
[Oct  2 16:43:54] NOTICE[4219]: chan_sip.c:3554 sip_call: SIPSRTP_CRYPTO
[Oct  2 16:43:54] NOTICE[4219]: chan_sip.c:3541 sip_call: SIPSRTP
    -- Called 1235
    -- SIP/1235-0a0c6258 is ringing
    -- SIP/1235-0a0c6258 is ringing
--- set_address_from_contact host '156.150.133.150'
[Oct  2 16:43:55] WARNING[4219]: sdp_crypto.c:161 sdp_crypto_activate:
Could not set remote SRTP policy
--- set_address_from_contact host '156.150.133.150'
    -- SIP/1235-0a0c6258 answered SIP/1236-0a0c0530
[Oct  2 16:43:56] NOTICE[4219]: rtp.c:3923 ast_rtp_bridge: Cannot native
bridge in SRTP.
[Oct  2 16:43:56] WARNING[4219]: rtp.c:1331 ast_rtcp_read: RTCP Read too
short
[Oct  2 16:43:56] WARNING[4219]: rtp.c:1624 ast_rtp_read: RTP Read error:
Success.  Hanging up.
  == Spawn extension (default, 1235, 3) exited non-zero on
'SIP/1236-0a0c0530' 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
10-02-07 10:06  kla960         Note Added: 0071319                          
======================================================================




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