How to deal with call quality issues


How to deal with call quality issues

Applicable to

All Cloud PBX Services users

VoIP call quality relies on the performance of your internet connection, router and local area network. Any delay, packet lost, jitter or variation on the internet signal can have an effect on voice calls.

Before opening a ticket with us, you can do a few things on your own

  • Test the audio quality or for latency.
  • Check our Status Page to see if there’s any current known issue that is affecting voice quality.
  • Run a VoIP test using our tool. Ideally, when the business is running and there is activity on the network to make sure it reflects the real life situation. You can then compare the result with another test done outside of working business hours. When asked by the test engine to enter a name, write it down so you can refer to it when opening a ticket.
  • Restart the ISP modem and router.
  • Contact the ISP and ask them to check the signal quality.
  • Configure Traffic Shaping/QoS in the router.

 If the problem is only with a specific extension:

  • Make sure you check the cabling, switch port, patch cables, and handset curly cord.
  • Make sure not to use a headset when troubleshooting call quality issues. Also, a wireless headset base needs to be a few inches away from the phone to prevent any interference.
  • If a computer is connected to the phone’s PC port, try disconnecting it and testing again. . An infected workstation can have an impact on the phone attached to the same cable.

When opening a ticket for call quality issues, make sure to include the following to help to speed up the process

  • VoIP Test results. Mention the name you entered while running the test.
  • Provide bad call samples, including date/time and phone number of each occurrence(s). (Eg: Extension 102 called 18195551234 at 2:32PM EST and there was static on the line, OR Ext 102 received a call from 15145558765 at 11:21 AM EST and the audio was choppy)
  • Provide any troubleshooting steps you made on the customer’s network.

 Requirements for VoIP

  • Ping should be under 150ms.
  • Jitter should be less than 3ms.
  • No packet loss should be present.
  • SIP-ALG should be off in the router configuration.
  • Each audio calls will use around 100kbps inbound and outbound

 A few things that can affect VoIP (mostly when traffic shaping/QoS is not setup)

  • Online backups done during working hours.
  • Remote VPN access / Branch office VPNs.
  • Remote camera monitoring.
  • DDOS attack on customer IP Address.
  • Cloud synchronization solutions like Google Drive, Dropbox, Onedrive, Apple Itunes phone sync.
  • Malware
  • Torrents


Jitter is the variation of delay between packets. High jitter will result in garbled voice.

Packet lost are packets sent and never received on the other end. It can be caused by too much traffic on the network, poor router quality, or dropped packets from ISP. In either case, it will result in missing voice packets (choppy audio).

Latency (ping) is the time it takes for a packet to navigate between source and destination. It will result in voice delays. After 150ms callers will start to notice the voice delay and it can become hard and confusing to have a live conversation.