[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Feb 27 08:19:22 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: 2011-02-27 08:19 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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(0132426) marqumch (reporter) - 2011-02-27 08:19
https://issues.asterisk.org/view.php?id=18674#c132426
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hi
i installed latest release of asterisk from svn ..
when imake a call on srtp .. it gives error...
when i try to make a call from eyebeam 1.5.7 it give follwoing error..
sip/sdp_crypto.c:226 sdp_crypto_process: Unsupported crypto parameters:
[Feb 27 09:11:50] WARNING[13888]: chan_sip.c:8409 process_sdp: Can't
provide secure audio requested in SDP offer
and when i try to make a call from phonerlite then it gives...
chan_sip.c:8414 process_sdp: We are requesting SRTP, but they responded
without it!
== Using SIP RTP CoS mark 5
i try to make call between two softphones... tls registration is working
well...
please help me out ...is this internal error or i do some mistake...
i need to make it work tomorrow..
thanks
Issue History
Date Modified Username Field Change
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2011-02-27 08:19 marqumch Note Added: 0132426
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