[OpenSIPS-Users] Question about opensips+asterisk
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?
> 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.
> 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
> was yes for the connection to asterisk (SIP protocol). I was
> 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
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
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