Hello, I tried to use mediaproxy, it includes two softwares (dispatcher & relay), I tried a lot to run more than one relay on the same server in order to bind them to different interfaces. But unfortunately this didn't work and I think it's not possible. I recommend using RTPProxy which is designed to work in bridging mode between two networks and you can run multiple instance of RTPProxy on the same server.
Regards. On Thu, 2009-08-20 at 15:48 -0400, Matthew S. Crocker wrote: > Can mediaproxy glue two RTP streams together (CallerA to CallerB)? > Can mediaproxy glue two RTP streams together from different interfaces/IPs > (eth0 & eth1) ? > > If so then it should be able to glue two calls together between public IP > (eth0) and private IP (eth1). > If the two RTP streams have to be on the same interface for mediaproxy to > work then I would expect to run into issues. > > EndUser <-> (eth0) MediaProxy (eth1) <-> SIP Gateway > > > ----- "Jeff Pyle" <[email protected]> wrote: > > > From: "Jeff Pyle" <[email protected]> > > To: "OpenSIPS users mailling list" <[email protected]> > > Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern > > Subject: Re: [OpenSIPS-Users] SIP Trunking > > > > 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 > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
