[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Feb 25 01:54:16 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18674
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Reported By: bbeers
Assigned To:
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Project: Asterisk
Issue ID: 18674
Category: Channels/chan_sip/SRTP
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA: SWP-3142
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 303637
Request Review:
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Date Submitted: 2011-01-25 09:56 CST
Last Modified: 2011-02-25 01:54 CST
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Summary: [patch] Unable to choose which SRTP suite to offer
Description:
Setting encryption=yes in sip.conf will cause asterisk to
generate a line in SIP INVITE SDP:
a=crypto: AES_CM_128_HMAC_SHA1_80 ...
There is no way to specify that asterisk should offer
AES_CM_128_HMAC_SHA1_32 instead of
AES_CM_128_HMAC_SHA1_80.
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Relationships ID Summary
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related to 0018187 Indicate SRTP + Feature reqest
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(0132398) Irontec (reporter) - 2011-02-25 01:54
https://issues.asterisk.org/view.php?id=18674#c132398
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Hi, I'm using Cisco SPA5XX phones and Snom360.
Without any patches and changing only the sdp_crypto.c to offer AES_32
instead AES_80 works well. We only have a Warning like:
WARNING[4187]: res_srtp.c:338 ast_srtp_unprotect: SRTP unprotect:
authentication failure
But, the conversation is protected (using Wireshark or Omnipeek we can't
hear real audio)
Change to AES_32 is required because SPA5XX phones send two Crypto lines
and the first one is AES_32 and as we know Asterisk only uses the first
line....
(With Snoms we can choose which type to use)
Thanks for your effort with this issue.
Issue History
Date Modified Username Field Change
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2011-02-25 01:54 Irontec Note Added: 0132398
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