[asterisk-dev] Paid support - Asterisk - Direct media - SRTP

[Listas" Nivaldo Montenegro Junior]

Hi,

So, It's not possible to implement using asterisk. The best solution is use
a SIP Proxy? Using a SIP Proxy the internal calls (exten to exten) will not
be processed by Asterisk. The SIP Proxy will handle that and Only calls
that goes to PSTN will be handled by Asterisk, right?

Regards,

Nivaldo Montenegro Júnior
Em 26/03/2014 03:48, "Olle E. Johansson" <oej at edvina.net> escreveu:

>
> On 26 Mar 2014, at 02:41, "Listas\" Nivaldo Montenegro Junior" <
> listasjr at digivox.com.br> <listasjr at digivox.com.br> wrote:
>
> Hi,
>
> We are looking for a developer or asterisk consultant to implement the
> direct media setup on Asterisk using TLS and SRTP.
> We tested using SIP + RTP and it works fine with the parameter
> directmedia=yes. But when we enable the SRTP, this stops to work.
>
> For SRTP to work peer to peer you need to have the key exchange either
> going through Asterisk or in the media stream, like DTLS. It may be easier
> to use a SIP proxy to get the key exchange to work properly.
>
> To facilitate SDES key exchange through Asterisk involves a lot of issues,
> which from a security standpoint are rather scary. Asterisk will offer
> someone else's keys and need to switch back and forth during the call.
>
> Anything can be done though, it's just source code. :-)
>
> /O
>
>
> We are very interested in this project and we will pay for it.
>
> If any one has interest on this project, please send me an e-mail to
> junior at digivox.com.br.
>
>
> Thanks,
>
>
> --
> *Nivaldo Montenegro Júnior*
> *Diretor de TI&C*
>
> *Digivox Soluções de Comunicação Ltda *( +  55 83 4009-8195
> ( +  55 83 4009-8199 - fax
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> :  junior at digivox.com.br
> : *www.digivox.com.br*
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