[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Feb 3 07:55:48 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: feedback
Asterisk Version: SVN
JIRA:
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: 2011-02-03 07:55 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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(0131437) bbeers (reporter) - 2011-02-03 07:55
https://issues.asterisk.org/view.php?id=18674#c131437
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c1nco: I still have something wrong in my flag translations
between the sip_pvt/sip_peer structure and the srtp structure.
I should be able to get that resolved today.
Here is evidence from your 'external caller to phonerlite client' trace:
"a=crypto:1 (null) inline:..."
and
[Feb 2 14:32:05] VERBOSE[23479] sip/sdp_crypto.c: == Selecting 0
((null)) for srtp crypto offer.
But, in your second trace, the INVITE from phonelite has
"m=audio 5063 RTP/AVP 0 8 111 110 97 101"
and asterisk complains because it expects to see RTP/SAVP.
That is a separate issue, IMHO, from the crypto suite patch.
Issue History
Date Modified Username Field Change
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2011-02-03 07:55 bbeers Note Added: 0131437
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