[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer

[Asterisk Bug Tracker]

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:                     new
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:              2011-01-25 12:50 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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 (0131038) bbeers (reporter) - 2011-01-25 12:50
 https://issues.asterisk.org/view.php?id=18674#c131038 
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I thought about that, but I wanted to avoid altering the existing behavior
 of the SIP_PAGE2_USE_SRTP flag as much as possible.  
Also, this is just a first step toward full SRTP compliance.  We need to
 be able to offer NULL cipher and F8_128_HMAC_SHA1_80 suite. And we need
 to be able to offer different suites for audio and video in the same
call.
 Trying to put all of that into a one bit flag will be difficult. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-25 12:50 bbeers         Note Added: 0131038                          
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