[OpenSIPS-Users] Question about opensips+asterisk

Bogdan-Andrei Iancu bogdan at voice-system.ro
Tue Oct 14 16:13:52 CEST 2008


A proxy cannot do that, so you have to go for client config. But 
Asterisk does not support SRTP (just asked Olle - seating next to me 
here ;) ).


Pierre astone wrote:
> By the way, how do you force the srtp protocol instead of the standard 
> rtp? is it just a client configuration, opensips configuration or 
> asterisk's? And if the srtp protocol is not supported by one of the 2 
> clients, is there a way to know it or does it just automatically 
> switch to rtp?
> Pierre
> On Mon, Oct 6, 2008 at 11:23 AM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>     Hi Pierre,
>     TLS is offering encryption only for signalling, so the media will
>     be still be vulnerable. For RTP part, there is SRTP (secure RTP)
>     which is end-2-end, so both devices (caller and callee) must
>     support it. AFAIK, SNOM phones support this.
>     Regards,
>     Bogdan
>     Pierre astone wrote:
>         Hi all,
>         Last time I asked if it was possible to use opensips to
>         encypher (via
>         TLS) an asterisk connection by using opensips as a proxy. The
>         answer
>         was yes for the connection to asterisk (SIP protocol). I was
>         wondering
>         if the voice conversation initiated via the SIP protocol is still
>         cyphered or if we have to find another way to cypher it. If
>         so, does
>         anyone have any idea on how to do so?
>         Thanks in advance
>         Pierre
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