Prepare Your Network for Phones (line test, ports and protocols)

Prepare Your Network for Phones (line test, ports and protocols)

Prerequisites

  • Active internet connection
  • laptop/desktop connected to the network being tested
  • access to the router/firewall  of the network

Router Compatibility

You can visit our router compatibility chart to see if there are any known issues or special configuration instructions for your model of router.

Readiness Tests

Ensure the readiness tests have been completed and the results are within required Parameters:

  • Line Test: https://linetest.iplogin.ca/

    Your results may vary but you should be looking for the number of supported calls to be higher than your requirements for the location and the ping and jitter should be "Great" or "Okay".



Router and Firewall Requirements

  • SIP-ALG should be disabled.

If the location has an aggressive firewall please allow all outbound traffic for the following ports to the following addresses. Our services use DNS-SRV and have many IP’s so the least restriction possible is important. 

Required Whitelist Addresses:

  • *.iplogin.ca (IP's available here)
  • time.google.com
  • pool.ntp.org
  • dm.yealink.com
  • 52.71.103.102
  • 35.156.148.166
  • 106.15.89.161
  • 47.75.58.202
  • 47.89.187.0

TCP Ports:

  • 80, 443, 8001 (for web UI)
  • 5060 (SIP)
  • 5061 (SIP TLS)
  • 9002 (Web Socket used for features like Web Phone and Portal Video)

UDP Ports:

  • 5060 (SIP)
  • 20000-27999 (RTP Media) - Inbound RTP Media
Outbound RTP ports utilize the entire unused port range on the internal network and should not be restricted. 

SIP-ALG Issues

SIP-ALG is a protocol used in VoIP telephony Services, A lot of routers have their own SIP-ALG enabled by default that will attempt to re-write SIP packets with their own information. SIP-ALG can and usually does exist on routers/firewalls and modems or in rare cases a managed switch. 

You must disable this feature for any VoIP service to operate as expected. Typical symptoms of SIP ALG being left turned on, include:

  • Phones dropping registration
  • Calls going straight to voicemail for no known reason
  • Calls dropping after a set period of time
  • Calls dropping when trying to retrieve the call from hold
  • Calls dropping when being transferred
  • Calls not being received (your phone doesn’t ring)
  • Unable to make outbound calls
  • Calls with no audio & Calls with one way audio
  • Other extensions continue ringing after a call has been answered
  • Call connectivity issues - a call is transferred but the person cannot hear the caller at the other end.

SIP-ALG Test

You can run a test for SIP-ALG here:
This test requires you to download a small utility called BCS the first time you use it. Download links are attached at the bottom of this article for your convenience.

http://mcstest.visualware.com/myspeed/myvoiph5_g711_sipalg.html




Your result should show that the SIP-ALG Firewall = N just like in the above picture.

If the test results show that you have SIP-ALG enabled you will need to go into your router configuration and disable it in your firewall settings, then save and reboot the router. Once the router reboots log back into the routers firewall settings and confirm SIP-ALG is disabled.


    • Related Articles

    • Line Test

      When troubleshooting issues with your phones it is crucial that a Line Test is performed to assist in identifying network issues which could affect your Quality of Service. Using Line Test Navigate to the Line Test Website and once there enter your ...
    • Setup Shared Line Appearances

      Prerequisites Office Manager or Higher Scope Available lettered extensions to be used Setup Notes Each device must have their own extension (usually will be a lettered extension) Will not work in a Call Queue currently Some phones may require ...
    • Network Assessment Instructions

      Prerequisites Windows 10 Computer that is on the same network as the phones Software license key What does this test do? This test allows you to perform a variety of tests on your clients sites, some of the tests include: Network assessment which ...
    • Cisco Meraki

      Current Status Compatible Suggestions None Comments Meraki VoIP Best Practices Generally speaking, it is best to stick to the following best practices when deploying a VoIP system on any network: Segregate voice traffic to its own VLAN. Voice traffic ...
    • 3CX SIP Trunk Setup

      Tested Version - 16.0.3.676 on July 9, 2020 NOTE - DNS-SRV on 3CX versions 15.5.15502.6 and above correctly failover when the first DNS entry is unreachable, but the 3CX system will not recheck the first entry until the second DNS entry is ...