[asterisk-bugs] [Asterisk 0014900]: [SRTP branch] RTP Read error: Success: Hanging up. If res_timing_pthread.so is loaded

[Asterisk Bug Tracker]

The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=14900 
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Reported By:                vrban
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14900
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 suspended
Fixed in Version:           
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Date Submitted:             2009-04-14 16:08 CDT
Last Modified:              2009-04-15 14:51 CDT
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Summary:                    [SRTP branch] RTP Read error: Success: Hanging up.
If res_timing_pthread.so is loaded
Description: 
Phone for this test: snom 320 FW_7.3.14

If the res_timing_pthread.so is loaded, rtp.c hangup the channel
beacause:

[Apr 14 22:40:50] DEBUG[29841]: res_srtp.c:301 res_srtp_unprotect: SRTP
unprotect: authentication failure
WARNING[29727]: rtp.c:1723 ast_rtp_read: RTP Read error: Success. Hanging
up.

But this i a bit hasty, because this happen only with the very first frame
that is comming from the phone. All following ones are ok. (tested this by
replacing return NULL; with return &ast_null_frame;)

Then you get the warning only once, and the srtp audio is ok between
ast<->snom

UPDATE:
this issue has it's origin probably here:
res_srtp_unprotect: SRTP unprotect: authentication failure

and again only the very first frame from the snom phone caused
the "RTP unprotect: authentication failure"

so there are two options, let the ast_rtp_read function from rtp.c be a
bit more relaxed, and hangup only if there are more the one bad frames from
the phone consecutively and/or let snom check if they do something wrong
with there first srtp packet they send to asterisk. 

this issue has been also reported in 14649 by me. But was closed because i
did a formal error using this bug tracker. if i do again a formal error,
then please correct it instead of just closing the report please.

this report is certainly for twilson

p.s. a category Channels/chan_sip/SRTP would be useful 
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---------------------------------------------------------------------- 
 (0103282) file (administrator) - 2009-04-15 14:51
 http://bugs.digium.com/view.php?id=14900#c103282 
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As I was mentioned in the other issue please report this problem as a note
on the original SRTP issue. We don't accept issue reports for a branch that
is still in progress in another issue like this. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-15 14:51 file           Note Added: 0103282                          
2009-04-15 14:51 file           Status                   new => resolved     
2009-04-15 14:51 file           Resolution               open => suspended   
2009-04-15 14:51 file           Assigned To               => file            
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