The subject been brought up before, but I thought I would try and bring it back with the MT eng team again. With NAT UDP sessions for VOIP and probably other stuff is broken. Even doing a bandwidth test sending to a MT router UDP data seems to be broken. Is there going to be a fix for this or are we going to have to give up on MT for NAT? We currently have two routers, one does NAT across the router with many sessions. Calls originating outside going inside work. Calls from the inside to the outside, originating from our customer do not. There is no ring tone heard and conversations are only heard from the inside to the outside. When the call originates from the outside all is well. The fact that the bandwidth test fails for udp traffic when sent seems like it must be related...

HELP

We run thousands of customers of VoIP through hundreds of ROS routers. Yes, it requires a great deal of knowledge of IP, TCP, UDP, NAT, SIP, RTSP, SDP, and other protocols, but the NAT functions of ROS are not broken. I will tell you that you should not use the SIP NAT helper as it's not good enough at this time and will cause you problems. I'll also tell you that one of your SIP devices should be on a public IP or at least using a one-to-one NAT. Two SIP clients behind a cone-NAT will not work without special NAT fixups.

So what are the special NAT fixups? We must use NAT across most of our customers....

One to one NAT does not save IP address space.