Well, not exactly. sipXbridge with a properly configured firewall will anchor the media. It will manage the RTP and the media relay.
RTP is plainly defined (default udp 30000-31000) and the symettric port nat which sipxmediarelay employs when supporting remote users handles the first call to be out:in udp 30000:30000, next call 30001:30001, etc. I thbik it you have remote workers enabled and server behind nat configured "it should work". Right now sipx is being told to route to ports it is not employing because the sipxbridge component is not being used not is the remote workers aspect. If you support either, you define an SBC, you have none. That's what sipxbridge does. Ingate makes a nice one too. On Thu, May 20, 2010 at 11:43 AM, Irena Dolovčak <irena.dolov...@gmail.com>wrote: > Tony, thank you for your answer. > > I understand the problem of NAT. The NAT is symmetric, so I thought I could > translate ports to internal IP addresses. (That's what I have done with 3CX > PBX and it worked). I can see the public IP address of the remote phone in > my registration in Sipx PBX: <sip:2...@81.193.210.15:5062 > ;line=3tg84a2k;x-sipX-privcontact=192.168.1.244%3A5062> > So here are the public and private ip addresses of the external phone. The > routers are configured that they translate specific ports to internal ip > address. The router knows the internal address and port from who he has > received the packet and he sends it back to the same ip address, at least it > should send it. > Correct me if I'm wrong, but I think that the phone wouldn't even register > if this, what I try to accomplish, can't be done. > > The Mikrotik can route the packets based on the RTP port, as long as all > phones use distinct ports for RTP. Another problem in my theory would be > the fact that the phone in my office tries to send the media through the > SipX and not to communicate direct to the other phone. > (I think that the phone should support Re-Invite, or am I wrong?) > Shouldn't the sipX at least support By-pass mode? > > I have already done the same thing with the 3CX PBX. In that case I also > had the 3CX PBX and one phone on the same subnet. Than, I had 2 more phones > on the external office. I managed it to work with no VPN, just through > double NAT with Mikrotik routers. > > > > > > Unless you have a a new trick up your sleeve that the world does not know >> about, I don't see how ANY sip server behind nat will handle media for a >> remote worker. >> >> > I don't even want that the sip server handles the media, I just want that > the server handles SIP, the media should go direct from one phone to > another. > > >> Your DNS (dyndns or not) needs to have all the records there, SRV and FQDN >> (A record). The A record on public DNS "has to point to the public IP of >> your Internet router". This is a HAVE-TO and not really optional or up for >> discussion in SIP. >> >> >> > Thanks, that's what's confused me because of the remote phone. > Probably a stupid question, but I couldn't fine any information about this, > what exactly are the SRV records for? Are they used for SIP Register or > also for RTP? > > > Regards, > > -- > Irena Dolovčak > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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