[asterisk-bugs] [Asterisk 0010024]: Adding ZRTP security protocol support for asterisk

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Jul 30 17:36:23 CDT 2007


The following issue has been RESOLVED. 
====================================================================== 
http://bugs.digium.com/view.php?id=10024 
====================================================================== 
Reported By:                sagarpai
Assigned To:                qwell
====================================================================== 
Project:                    Asterisk
Issue ID:                   10024
Category:                   Core/General
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     resolved
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!):  
Disclaimer on File?:        Yes 
Request Review:              
Resolution:                 suspended
Fixed in Version:           
====================================================================== 
Date Submitted:             06-20-2007 21:53 CDT
Last Modified:              07-30-2007 17:36 CDT
====================================================================== 
Summary:                    Adding ZRTP security protocol support for asterisk
Description: 
Support is to be added for handling ZRTP protocol ( www.zfoneproject.com)
in Asterisk. ZRTP is the security protocol that secures the RTP stream.
Philip Zimmermann is the primary designer of the protocol and has developed
the libzrtp library implementing the protocol.

The ZRTP-support will handle the following scenarios

1.There is a need to Monitor traffic, handle DTMF between a caller
(Authenticated user of PBX) & a callee (Authenticated user of PBX).

2.There is a need to Monitor traffic, handle DTMF between a caller
(Authenticated user of PBX) & a callee ( not an Authenticated user but on
RTP ).

3. There is a need to Monitor traffic, handle DTMF between a caller
(Authenticated user on PBX A) & a callee ( a Authenticated user on PBX B).

4.There is a need to Monitor traffic, handle DTMF between a caller
(Authenticated user of PBX) & a callee ( not a Authenticated user and not
on RTP ).

5.There is a need to Monitor traffic, handle DTMF between a caller
(Authenticated user of PBX) & a callee (a Authenticated user and not on RTP
).

6. When both peers have no Media Translation, DTMF handling needs
====================================================================== 

---------------------------------------------------------------------- 
 qwell - 07-30-07 17:36  
---------------------------------------------------------------------- 
My question from 3 weeks ago wasn't answered, and there has been no patch
uploaded, so I'm going to assume that this is a feature request.

Closing.  If you do have a patch for this, please feel free to reopen.

If it's a feature request, I would recommend putting up a bounty for it. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-30-07 17:36  qwell          Status                   new => resolved     
07-30-07 17:36  qwell          Resolution               open => suspended   
07-30-07 17:36  qwell          Assigned To               => qwell           
07-30-07 17:36  qwell          Note Added: 0068094                          
======================================================================




More information about the asterisk-bugs mailing list