> I have a question for you: > > - "SIP doesn't work behind NAT very well" > > Do you agree with this sentence?
Depends. Asterisk behind a nat box tends to be an implementation problem limited by the knowledge of the person doing the implementation and somewhat by the functionality implemented within the nat box. Sip phones behind a nat box (with asterisk on a registered IP address) tends to be rather easy, and how well it works depends a lot on how well the sip phone vendor implemented nat support. Both asterisk and sip phones behind different nat boxes tends to be the most difficult to implement and requires the greatest amount of knowledge/experience to implement. Again, depends a lot on the functionality provided in the nat boxes. The issue with sip is that session startup and control occurs across udp port 5060, and the two endpoints (* and phone) negotiate another set of udp ports for the rtp (voice) session. The choice of which rtp ports to use was left up to each sip phone vendor, so the udp port number in use could be anything from about 8000 (xlite) to something greater then 32,000. Some firewall/nat boxes will actually watch the sip rtp negotiation process by inspecting the contents of the sip packets, and open up the wanted ports. However, most cheap nat boxes don't do that, and leave it up to you to statically define/map the ports required. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
