[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Feb 28 05:54:56 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: 2011-02-28 05:54 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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(0132432) Irontec (reporter) - 2011-02-28 05:54
https://issues.asterisk.org/view.php?id=18674#c132432
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Hi bbeers,
You're right. This Warning appears when the phone sends to Asterisk a RTCP
packet with "Source Description" header.
I've been checking Asterisk code to find hard-coded about RTCP but no
luck...
And using Cisco phones I can't change AES_32 because the phones send this
mode as first crypto line, (they send both lines, AES_32 and AES_80) but we
can't choose between them.
Maybe Asterisk can't handle those packets? I mean, all RTCP packets from
phones to Asterisk have "Source Description" and Asterisk complains with
the Warning.
I don't know...
Issue History
Date Modified Username Field Change
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2011-02-28 05:54 Irontec Note Added: 0132432
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