The following check list offers some general settings that apply to most routers. These settings should create an optimal environment for VoIP interoperability.
- Disable Application Layer Gateway (AKA – “SIP Awareness”)
- Set UDP window timeout to 60 seconds
- Disable STUN, ICE or any other local NAT traversal settings
- Enable QoS for local devices. By IP address , By MAC address, By VLAN ID
- Enable Bandwidth QoS (if available)
- Set the upload speed of the internet connection.
Router will prioritize based on the available bandwidth.
- Create firewall policy if the local policy is blocking SIP and/or RTP. o Allow UDP port range 49152- 64512 o Allow UDP+TCP port 5060
If your router is connected to broadband MODEM supplied by your internet service provider, it is possible that some or all of the above settings should be set within the supplied device as many Modems act as firewall/router devices .
Port Forwarding (NAT) Policies for Flowroute's Direct Audio
To ensure you receive all audio on your Flowroute calls, specific Port Forwarding/NAT policies should be put in place on your network. The following two Port Forwarding network address translation (NAT) policies are required:
SIP signaling (call control): Forward UDP and TCP traffic on port 5060
 to your PBX's local IP address.
 RTP media (call audio): To reduce latency, Flowroute uses Direct Audio. To receive Direct Audio, allow UDP packets from any source IP address with a destination port within your system's RTP media port range; forward to your PBX's local IP address.
- If your system has issues connecting over port 5060, you can use 5160 as an alternate SIP port.
- If your Port Forwarding configuration allows you to specify the source IP of your SIP traffic, you can restrict traffic on port 5060 or 5160 to either of the following server IP addresses: 22.214.171.124 or 126.96.36.199.
- RTP media port range varies by phone system. Your system's RTP media port range will be configured locally on your system and/or detailed in your system documentation.