help:SRTP::minisip

[Alexandr Olekhnovich]

Hi,
really there is no SRTP support in asterisk (i mean mainstream) for the
moment as I know, but there is an SRTP solution by the third-party. I use it
for 1.4.
http://bugs.digium.com/view.php?id=5413
Take a look at the link above, here is a patch for 1.2 but there was no
problems for me to adapt for 1.4.
Guess it'll help ;)

On Jan 6, 2008 4:47 AM, Cesc Santa <cesc.santa at gmail.com> wrote:

> Let´s first involve all users and developers from the list ...
>
> ---------- Forwarded message ----------
> From: mehdi boumezzough < mehdi02000 at yahoo.fr>
> Date: Jan 5, 2008 5:35 PM
> Subject: help:SRTP::minisip
> To: cesc.santa at gmail.com
>
>
> hi
>
>  May I first introduce myself. I'm a student of ENSA
> MARRAKECH from morocco currently working for my
> professor on a VOIP project.I don't speak english very
> well but I want to try.I was interested in securing
> the RTP stream in asterisk wiht SRTP and
> minisip.Please i'd be extremely
> grateful for any document that can help me.
>
> Thanks in advance,
>
> M.BOUMEZZOUGH
>
>
>
>
>
>
>
>
>
>
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-- 
Best Regards
Alexander Olekhnovich
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