Good Morning, I have an existing SIP-based hosted PBX service running on a commercial platform. To handle some edge cases, I would like to setup a proxy to achieve two purposes: transcoding from iLBC, SILK & G.722 to G.711 and offering SIP over TLS and SRTP to customers that require it. I've never used OpenSIPS so I'm starting the learning process from scratch.
Some key points I have found, so far: 1. Since the proxy will be transcoding and/or handling SRTP streams, it will also need to handle NAT traversal; 2. It should be completely stateless, I think; 3. UACs will be registering through it (they will be configured with it as an outbound proxy); 4. If one UAC ends up being connected to another UAC (e.g. extension to extension dialling), we want the SRTP stream to simply be relayed; 5. This proxy will simply relay all signalling and decrypted/transcoded media to the existing switch; It would be great to hear all your comments and suggestions of what components and modules to use and any other tidbits you can provide. Also, if anyone has done something like this already and wants to offer commercial services to replicate it for me, send me an email. Thanks, A. -- Adam Sherman Technologist Deputy SARCOM, SAR Global 1 Coordonnateur de L'AQBRS, région 07 Outaouais +1 613 797 6819 _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
