[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:              2011-02-07 10:15 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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 (0131564) bbeers (reporter) - 2011-02-07 10:15
 https://issues.asterisk.org/view.php?id=18674#c131564 
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c1nco:  Looks like we're back where we were before I caved to
 the "no new parameters" suggestions.
The SRTP unprotect warnings are happenning at 5 second intervals.  
Could they be (S)RTCP related?  
( I have not had any luck using (S)RTCP while using SRTP.  
  In fact, one of my other, related, patches is to ignore incoming
  (S)RTCP packets when using SRTP. Another issue to bring up soon.)  
Can you disable (S)RTCP from the other end and see if those
 warnings stop?

As for the Phonerlite trace, the problem is the same as
 before ...  Asterisk sees the a=media line:

m=audio 5062 RTP/AVP 111 110 97 3 0 8 101

And decides that secure_audio is off.
I can't quote RFC chapter and verse, but Asterisk expects
 it to be:

m=audio 5062 RTP/SAVP 111 110 97 3 0 8 101 

You could modify the logic in process_sdp() function in chan_sip.c,
 but I have no idea of the repercussions of such a decision. 
I think the other end should be fixed. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-07 10:15 bbeers         Note Added: 0131564                          
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