[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:                     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:              2011-02-25 07:58 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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 (0132405) bbeers (reporter) - 2011-02-25 07:58
 https://issues.asterisk.org/view.php?id=18674#c132405 
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Irontec:  IMO, that 5 second warning must be RTCP related.  
 Your wireshark trace should show the correlation between the
 warnings and the arrival of packets on the RTCP_PORT.
Does it only happen when using AES_32?  
Several people complain about it after trying to use aes_32. 
There may be some disconnect between how RTP and SRTP get "unprotected"
 inside Asterisk.  (I wonder if SRTCP is somehow 'hard-coded' to aes_80?) 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-25 07:58 bbeers         Note Added: 0132405                          
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