[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:                     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:              2011-02-01 17:52 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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 (0131373) c1nco (reporter) - 2011-02-01 17:52
 https://issues.asterisk.org/view.php?id=18674#c131373 
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Here is some logs from an external call to phonerlite client

http://pastebin.com/Sarg21fL


Here is Phonerlite when trying to dial VM box

http://pastebin.com/ysWQTMNv


I was able to connect to phonerlite client through external call (cell
phone) on your previous patch (rev 2) however asterisk would spam: SRTP
unprotect: authentication failure

I now cant connect to any client at all.

I using the below variable per extension 201
encryption=aes_32

Any thoughts? 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-01 17:52 c1nco          Note Added: 0131373                          
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