[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.
======================================================================
https://issues.asterisk.org/view.php?id=17563
======================================================================
Reported By: Alexcr
Assigned To: twilson
======================================================================
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:
======================================================================
Date Submitted: 2010-06-29 04:04 CDT
Last Modified: 2010-09-13 02:36 CDT
======================================================================
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 ...
======================================================================
----------------------------------------------------------------------
(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
======================================================================
More information about the asterisk-bugs
mailing list