RTP can use any ports... it needs 2 ports the first one on an even port and the next port up from that. One port is used for the audio, the other for control. Asterisk is setup to accept 10000 RTP ports from 10000-19999. You need to look in your 286 setup and see what ports it's set to use, probably around 16384.
SIP has nat issues because the audio RTP is separated from the SIP signaling. This is usually only a problem when you have a phone call another phone when their both behind nats. If you have a server in between them like an Asterisk, the server with help translate the call past the nat. Teliax uses Asterisk as the server to give you PSTN connectivity. They should have set Asterisk to do nat traversal so you don't need to set anything for your 286. Basically look at your account and look on the support page. The Linksys PAP2 is similar in setup for their service. If you work with teliax on your issue again, tell them your setting up your ATA for their service and they need all the nat=yes and other setting for nat in their sip.conf to fix you problem. IAX uses only one port for signaling and audio on 4569. Its also nat friendly and almost always works on a nat without and configuration on the router. On Nov 26, 2007 3:16 PM, <[EMAIL PROTECTED]> wrote: > On Mon, Nov 26, 2007 at 02:46:47PM -0800, Gil Lamb wrote: > > SIP uses port 5060 for signaling and RTP for the audio. > > I believe you but now I'm confused because Teliax support said SIP > needed all of ports 10,000 thru 20,000 open. I thought that > sounded a little excessive. > > What port does RTP use? > > > Chris > > > -- > [email protected] > http://www.kernel-panic.org/cgi-bin/mailman/listinfo/kplug-list > -- [email protected] http://www.kernel-panic.org/cgi-bin/mailman/listinfo/kplug-list
