[asterisk-bugs] [Asterisk 0017563]: [patch] SRTP (SRTP unprotect: authentication failure)

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Sep 13 02:36:23 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17563 
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Reported By:                Alexcr
Assigned To:                twilson
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Project:                    Asterisk
Issue ID:                   17563
Category:                   Resources/res_srtp
Reproducibility:            always
Severity:                   block
Priority:                   high
Status:                     assigned
Target Version:             1.8.0
Asterisk Version:           SVN 
JIRA:                       SWP-1795 
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/878/ 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-06-29 04:04 CDT
Last Modified:              2010-09-13 02:36 CDT
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Summary:                    [patch] SRTP (SRTP unprotect: authentication
failure)
Description: 
I setup asterisk-trunk with res_srtp 

Test with Snom320 and Eyalink T26P is failed with error

[Jun 24 17:08:08] DEBUG[629] res_srtp.c: SRTP unprotect: authentication
failure
[Jun 24 17:08:08] WARNING[629] res_rtp_asterisk.c: RTCP Read error:
Success.  Hanging up.

with eyebeam-1.5 all work fine if add TLS support on both.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0017855 Asterisk with SRTP Record and Playback ...
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---------------------------------------------------------------------- 
 (0126866) marcelloceschia (reporter) - 2010-09-13 02:36
 https://issues.asterisk.org/view.php?id=17563#c126866 
---------------------------------------------------------------------- 
after an phone uptime of 21 hours, i will get:

  == Using SIP RTP CoS mark 5
    -- Executing [600 at from-sip:1] Playback("SIP/snom-00000005",
"demo-echotest") in new stack
[Sep 13 09:30:02] WARNING[14147]: res_rtp_asterisk.c:1642 ast_rtcp_read:
RTCP Read error: Success.  Hanging up.
  == Spawn extension (from-sip, 600, 1) exited non-zero on
'SIP/snom-00000005'
    -- Incoming call: Got SIP response 403 "Use Proxy" back from
172.17.3.101:2048


wireshark dump attached 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-13 02:36 marcelloceschiaNote Added: 0126866                          
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