[asterisk-dev] SRTP Status and Plans for Asterisk 1.6.x

[John Todd]

On Nov 9, 2008, at 8:24 AM, Stefan Tichy wrote:

> I applied the Secure RTP patch from bugtracker ID 0005413 to
> asterisk 1.6.1-beta1. Basically it works for calls from one phone to
> another, but there has been no development on this patch for some
> time.
>
> Are there plans to add SRTP to astersik with some 1.6.x Version?
> I think there was some information on that issue available but I am
> not shure and cannot find it any more.
>
> Thanks.
>
> --  
> Stefan Tichy  ( asterisk2 at pi4tel dot de )



I can't say what the plans are (as often, contributors are the source  
of "plans" with good code, such as yourself.)  Of course, there is a  
desire by many people to see SRTP in Asterisk - we've been lacking  
secure media communications for too long.  To my eye, it appears that  
we have overcome the most basic of problems that had previously  
prevented this patch from being applied, which was the lack of a  
secure key exchange mechanism.  Now that SIP/TLS is in place, perhaps  
that problem is no longer the primary obstacle.

Looking at the ticket in Mantis (http://bugs.digium.com/view.php?id=0005413 
) it does seem that there are compilation and patch issues, and even  
some compatibility problems.  If you have it working in a stable  
environment,  your assistance with getting this patch into shape for  
TRUNK submission might be all that is required to make it happen!   
This feature patch has been one of the longest outstanding in the bug  
tracker; hopefully it can be committed soon after more review and  
repair by good coders like yourself.

JT

---
John Todd
jtodd at digium.com        +1-256-428-6083
Asterisk Open Source Community Director