[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Feb 18 13:51:31 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: 2011-02-18 13:51 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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(0132161) bbeers (reporter) - 2011-02-18 13:51
https://issues.asterisk.org/view.php?id=18674#c132161
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andrewhack: I am deep into porting some other patches forward to
1.8.2.3, and can't dig into what is going wrong with
srtp-cryptosuite-bbeers22.patch today.
All I have really looked at is the construction of the INVITE/OK
SDP a=crypto lines (which look ok).
I hope I will be able to test the actual SRTP stream early next
week with Asterisk 1.8.2.3. Look for something (an updated
patch) on Tuesday or Wednesday (/me fingers crossed).
In the meanwhile, if anyone else can see what is amiss, please share.
BTW, srtpcapable is not necessary in Asterisk 1.8.x, AFAICT.
Issue History
Date Modified Username Field Change
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2011-02-18 13:51 bbeers Note Added: 0132161
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