These can be a real hurdle when setting up VoIP, help others by posting your configs, tips and tricks or simply ask others for help if you're stuck.
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By GoofyCyborg
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Most SIP-based VoIP services require specific dedicated ports to establish communication links between the VoIP Proxy Server and your VoIP User Agent (softphone or ATA adapter).

The default SIP port (Session Initiation Protocol) for setting up a SIP-based VoIP session is port 5060. Your NAT Rrouter and PC firewalls must open port 5060 (UDP) at a minimum. But, it doesn't end here. More ports are usually involved.

SIP Port 5060 is required to "setup" the links between you and the proxy server. But, the actual audio packets typically require other ports, as well, for carrying on a voice conversation.

The other ports required are the RTP (Real-time Transport Protocol) ports, which carry the actual data of the voice conversation. These ports are typically negotiated between the VoIP proxy server and the user agent (soft-phone/ATA).

For example, most SIP based VoIP services will negotiate with your device to allocate an RTP port, or ports, in the "range" between UDP ports 10,000 and 20,000. More specifically, my Linksys PAP2T-NA and SPA2102-NA ATA defaults to a required port range of between 16,384 - 16,482 to be open through the firewall to carry its RTP data.

So, the key for you, the user, is to know how to setup and verify that your NAT Router Firewall is configured to "Port Forward", or "Port Trigger" SIP port 5060 (and sometimes 5060 - 5070) as well as RTP ports 10,000 - 20,000, or whatever the specific range your soft-phone or ATA devices default to and require.

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