Issues with call quality

Issues with call quality

  • Checklist
  • Network diagnostics
  • Technical requirements
  • Global recommendations on call quality
  • VoIP (voice over IP) itself and Voiptime Cloud solution performance are both very sensitive to the underlying network and bandwidth. Inconsistency of the bandwidth can cause poor VoIP connection and, in such a way, problems with call quality.

    VoIP call quality depends mostly on your Internet connection, local infrastructure, network settings, and headset.

    Please use this short checklist to allocate the reason for the existing call quality issues.



    • The issue is irregular, only with some calls - there can be issues on the callee side (the cell phone is in the bad zone, the carrier used the unstable route, etc.)
    • All calls have the same issue - it is the reason for further investigation.

    Types of audio issues:

    • You and the callee couldn’t hear each other - please give us an exact description of such situation (when and to what phone number you tried to call, attach call record, too).
    • Only one side can be heard - please contact your system administrator to check the NAT and firewall settings.
    • Audio quality is bad - check the sound card, microphone and volume settings. Try to make the call from another computer with another headset.
    • Calls are dropped or there was a long delay between the phrases - in most cases the main reason is the usage of Wi-Fi connection*. Also, check your router settings and the stability of the internet connection.

    *Wi-Fi connection is not recommended for VoIP telephony because of its instability. We recommend the usage of wired internet connection only, with the minimal speed 100 kbit for each call. With the usage of WebRTC codecs this parameter can be decreased up to the 40-50 kbit for a call.

    If none of the suggested decisions helped, please make the network diagnostics as is described below and attach results to the ticket.

    Network diagnostics

    1. Start the network diagnostics using PingPlotter tool
    2. Run .exe file and follow the setup wizard.
    3. Run the PingPlotter and enter as IP, and change the interval to 1 second, then press “Start” button.
    4. Keep running for 15-20 minutes approximately.
    5. Export the received report via the “File” -> “Export Sample set” - “Data in Current Focus” menu option, select .pp2 file type.
    6. Send it to us.

    Please run the test in the standard working situation (you shouldn’t close opened tabs and programs you use).

    If you use Linux, run such command in the console:
    mtr -s 1000 -r -c 1000 -o “LDRSNBAWVGJMXI”

    Technical requirements


    Parameters Requirements
    Packet loss 0% packet loss
    Jitter less than 20 ms jitter
    Network latency less than 100 ms latency to Voiptime Cloud data center. VoIP services are known to work even in higher latency conditions up to 150-200 milliseconds. However, this must be maintained consistently with no packet loss.
    Internet connections from outside the United States are subject to higher latency, which can affect the Quality of Service (QoS).
    Bandwidth requirement G711 Codec: 90 kbps symmetric/call
    G722 Codec: 90 kbps symmetric/call
    Please make sure you have 50% of your available bandwidth free to accommodate any spike in usage.
    Always assume that at least 35% of your users are on call at any time. However, depending on your company’s use case, you may have a higher percentage.
    IP and RTP Traffic Do not restrict SIP and RTP traffic across LANs and/or WANs when multiple customer sites exist.

    If you are interested in QoS and want to prioritize voice traffic, please contact our IT team and describe your current infrastructure.

    QoS is highly complicated since it presupposes a lot of requirements both on our side and your WAN and LAN connections. To meet these requirements you should do the following: traffic marking, traffic prioritizing, etc. Should you be interested in fulfilling the above-mentioned requirements, we need to work on this together with our IT department.

    TCP / UDP Port requirements

    Port TCP / UDP Outbound Description
    443 TCP Inbound / Outbound
    Web application (https)
    7443 TCP Inbound / Outbound
    Internal communication for WebRTC phone
    5080 UDP Outbound
    Connection with the external VoIP carrier

    Recommended OS and browsers

    • Windows - 7 or newer
    • Google Chrome - Most recent stable version

    Global recommendations on call quality

    To improve the call quality please follow the recommendations below:

    • Use high-quality USB headsets for your agents
    • Get the latest updates for sound card and USB headsets drivers
    • Ensure there is no conflict with antivirus applications
    • Avoid video-streaming or high-volume download while the VoIP call is in progress
    • Ensure the recommended bandwidth requirements:
      • use wired internet connection for all your agents
      • 50% of your available bandwidth is free to accommodate any spike in usage
      • at least 35% of your users are on call at any time
      • a call requires about 90 kbps symmetric/call.

    Run the network test using PingPlotter and send us the results to check the situation with your current network settings.