[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.
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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: 10-02-2007 10:06 CDT
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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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kla960 - 10-02-07 10:06
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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
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10-02-07 10:06 kla960 Note Added: 0071319
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