3rd Party Firewall Rules: Difference between revisions

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=== Allow ===
=== Allow ===
<blockquote>
<blockquote>
==== SSL ====
==== HTTP ====
</blockquote><blockquote>
==== Secure HTTPS ====
</blockquote><blockquote>


* '''TCP, out:''' 80, 8080


==== HTTPS/SSL ====
* '''TCP, out:''' 443, 8443


</blockquote>
</blockquote>

Revision as of 21:58, 5 March 2024

This page is part of the Network and QoS guides.

Networking

Step one would be, can you see our servers? You can check here: https://core1-dal.vestednetworks.com/webtest/webtest.php

After you have confirmed that our servers are reachable via HTTPS/SSL ports, you will need to make sure some more VoIP specific ports are open on your network.

IP Addresses

The easiest way is to allow any ports to and from the following IP addresses to pass through your firewall.

If you cannot open all ports to the following addresses, you must open the ports listed below to the following IPs .

Allow

Voice/Video Traffic

  • Primary Addresses: 216.58.152.240/28
  • Secondary Addresses: 74.63.180.0/28

Fax Traffic

  • ATA: ataserver.ipfax.net & ataini.ipfax.net

Voice/Video Ports

If you cannot open all ports to an IP address, you may need to open specific ports through your firewall.

The following ports must be allowed to pass through your firewall to ensure proper function of your phone systems.

Allow

SIP (SIP Handshaking)

  • UDP/TCP, in/out: 5060-5062

WebSockets (NOVA Web Softphone, Mobile App)

  • TCP, in/out: 9002

RTP (Streaming Audio and Video for calls)

  • UDP/TCP, in/out: 20000-32000

Fax Ports

The following ports must be allowed to pass through your firewall to connect to the web portal and to use faxing.

Allow

HTTP

  • TCP, out: 80, 8080

HTTPS/SSL

  • TCP, out: 443, 8443

Advanced Routing Options

Not all firewalls, gateways, or modems will have these options visible. You may have to contact your ISP in order to make sure they have these options disabled on your incoming connection.

Disable

Settings

  • SIP-ALG
  • H.225

Sonic Wall Specific Routing

Disable

Settings

  • SIP Transformations

Enable

Settings

  • Consistent NAT
  • UDP Timeout to 120

VoIP Bandwidth Requirements

Our default compression uses 90kbps up & down for one single call.

Example: 6 active calls is roughly 540kbps up & down reserved. This would be a normal use case for ~20 seats with an average use of 30%.