Hi there, as a completion - as OpenSIPs does not directly do media, it cannot convert SRTP <-> RTP. Also, AFAIK, there are no decent tools (I mean only RTP tools that can be driven by OpenSIPS) to do this kind of translation. But never now what may come up in the future ;)
Regards, Bogdan Iñaki Baz Castillo wrote: > 2009/3/27 Serge Berney <s.ber...@kinonline.net>: > > >> I’m new user of OpenSIPS and desire to make this kind of configuration : >> >> >> >> IPPhone ß SIP TLS & SRTP à OpenSIPS ß SIP & RTP à Asterisk >> >> >> >> (Asterisk is working well and have CDR/User AUTH & Dialplan) >> >> OpenSIPS is well compiled and seems to work J >> >> >> >> But now, which module must I enable on OpenSIPS to enable TLS, >> > > Please look at OpenSIPS wiki for TLS tutorial. > > > >> SRTP and to redirect traffic to Asterisk ? >> > > OpenSIPS is a SIP proxy, it doesn't handle media. > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
