I've had three VoIP providers and two different home routers and have never needed port forwarding.
Anyone know why some consumer routers need port forwarding declared for VoIP ATAs and others don't?
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I've had three VoIP providers and two different home routers and have never needed port forwarding.
Anyone know why some consumer routers need port forwarding declared for VoIP ATAs and others don't?
VOIPo does not typically proxy its own audio. This means that the RTP stream will usually come from a different IP address than the SIP proxy which was used to set up the call. Some routers, particularly those with symmetric NAT, will drop packets which are sent to the correct address and port, but from a different IP address. That's at least one explanation. I'm sure there are others as well.
I can reverse the question ... I've had several VoIP providers and have never had to use port forwarding with my ancient router. Recently, I've started having a variety of issues and have started experimenting with port forwarding. So I guess, my question would be why my router works perfectly well with other providers but appears to have an issue with VOIPo.Quote:
Originally Posted by stevech
Perhaps because those other providers proxy their own audio. The explanation I gave above links the problem to both the router AND the provider.Quote:
Originally Posted by Russell
Russell:
This is a really great summary related to your question that I borrowed from wikipedia:
Quote:
Some Application Layer protocols (such as FTP and SIP) send explicit network addresses within their application data. FTP in active mode, for example, uses separate connections for control traffic (commands) and for data traffic (file contents). When requesting a file transfer, the host making the request identifies the corresponding data connection by its network layer and transport layer addresses. If the host making the request lies behind a simple NAT firewall, the translation of the IP address and/or TCP port number makes the information received by the server invalid. The Session Initiation Protocol (SIP) controls Voice over IP (VoIP) communications and suffers the same problem. SIP may use multiple ports to set up a connection and transmit voice stream via RTP. IP addresses and port numbers are encoded in the payload data and must be known prior to the traversal of NATs. Without special techniques, such as STUN, NAT behavior is unpredictable and communications may fail.
What he said. :)