Matthew, While I'm no Mediaproxy expert, I have seen many conversations on this list where Mediaproxy is described as a part of a far-end NAT solution. It was not designed to have a private IP attached to it. For that, you most likely will want to look at the rtpproxy application.
It sounds like you are constructing a local ALG to connect private and public networks. You don't necessarily need a full-blown Acme for that. I've had great luck with Edgewater Networks' "Edgemarc" devices, for example. That's just one. There are many. - Jeff On 8/20/09 2:49 PM, "Matthew S. Crocker" <[email protected]> wrote: > > I understand that OpenSIPS is not a full blown SBC (I can't afford an > ACMEPacket). Will it perform the functions to proxy the SIP & RTP streams > (via mediaproxy) between my end users and my internal gateway? > > At some point I plan on increasing the use of openSIPS to handle registration, > presence, routing, etc. > > -Matt > > ----- "Alex Balashov" <[email protected]> wrote: > >> From: "Alex Balashov" <[email protected]> >> To: "OpenSIPS users mailling list" <[email protected]> >> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern >> Subject: Re: [OpenSIPS-Users] SIP Trunking >> >> Matthew, >> >> Look for the mediaproxy module. >> >> That said, do be aware that a proxy is, by definition, not like an >> SBC. >> SBCs have many other capabilities a proxy does not; a proxy is a >> relatively "thin" interoperation layer. >> >> Perhaps the recently introduced b2bua module is brought to bear on >> that >> somewhat, but classically, OpenSIPS is a proxy. >> >> -- Alex >> >> Matthew S. Crocker wrote: >> >>> Hello, >>> >>> I'm brand new to OpenSIPS, just going through the make process now. >> >>> >>> I need to configure OpenSIPS to act like a SBC for some SIP trunks >> coming off a VoIP switch. Where should I look for >> Documentation/Examples of a working config? >>> >>> Here is my scenario: >>> >>> OpenSIPS has two interfaces, private & public. >>> VoIP Gateway is on private LAN with no gateway configured (it can >> only talk to local machines, no routing) >>> >>> End user has an Asterisk server on a private lan behind their >> firewall (NAT) >>> >>> I need to configure OpenSIPS to listen for SIP messages on :5060 >> from the end user firewall. It then need to rewrite the SIP message >> and send it to the Gateway. The Gateway would see the messages coming >> from the internal IP of the OpenSIPS server. Once all of the SIP >> messages get processed I then need the OpenSIPS server to proxy the >> RTP streams (plan on using mediaproxy) between the Asterisk server and >> VoIP Gateway. >>> >>> Any helpful hints on where to look? >>> >>> -Matt >>> >>> >> >> >> -- >> Alex Balashov - Principal >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
