Our system is designed so on the client-side the apps and IP Handsets establish the connection with the backend automatically without the need for any changes when using a standard ISP modem.
In some cases where the modem's firewall is a bit more restrictive or when you have a custom firewall, you may want to check the settings below for VoIP to work well.
- Make sure all outbound and inbound traffic is allowed to the following IP range where our infrastructure resides: 185.87.180.0/22
- Depending on the IP handsets configuration, IP handsets can use the following protocols for SIP signalling:
- TCP port 5060
- TLS over TCP port 5061
- TLS over TCP port 443.
- UDP port 5060
- The audio traffic is transmitted bi-directionally over UDP on any of the ports between 10,000 - 30,000.
For NAT traversal we use periodic keep-alive packets that are sent from the client device to the cloud server at a fixed interval to keep the NAT session open for incoming traffic from our servers. Please make sure the TCP and UDP NAT session timeout are not less than 60 sec.
In most case, you will also want to disable the firewall feature called SIP ALG or SIP Passthrough.