Identifying common call quality issues

When it comes to call quality, there are many variables and factors that can cause disruption. To determine the quality of your internet connection, you need to look into several parameters besides your download and upload speed. To see if your connection is up to the task, make sure to pay attention to the following:

• Bandwidth
• Latency
• Packet loss
• Jitter

Missed incoming calls

This can happen if your router closes connection by inactivity timeout. To fix this problem you need to reduce the timeouts in your account settings. With one of these minimum values the problem is commonly solved: "Keep-Alive" = 10 or "Register Refresh" = 150 on MicroSip.

We do not recommend setting the "Register Refresh" value too low. Use minimum value 150.

Audio is cutting out

You might be experiencing packet loss. When one or more packets of data traveling across a network fail to reach their destination, it will cause lost data transmission bits. This needs to be 0, usually expressed as a percentage (0%).

There is a voice delay

You might be experiencing latency. Latency - sometimes referred to as "lag," which causes interruptions—similar to lag in video games or video streaming. Latency is measured using the ping time, which is the amount of time the system takes to open and establish the connection. Ping time is measured in milliseconds (ms), and 100ms is considered as excellent, while anything over 250ms is considered poor.

Generalized degraded quality

You may be experiencing packet loss, latency, or jitter. Jitter, which is slightly different from latency, is a variation in delayed of received packets. Jitter and latency can be caused by network congestion, wireless network connections, and old/damaged hardware.

VoIP relies on the internet to transmit voice data, and any issues with the internet connection can result in poor call quality, dropped calls, or other problems. Here are some reasons why testing the internet connection is important for VoIP:

• VoIP requires a certain amount of bandwidth to transmit voice data without interruption. Each concurrent call needs to have 100 kbit/s of bandwidth available (both upstream and downstream).

• Latency, or the delay in transmitting data over the internet, can affect the quality of calls. Testing the internet connection can help determine if latency is within an acceptable range to make and receive calls.

• Jitter refers to variations in the delay of data transmission over the internet. High jitter can result in choppy, distorted, or delayed audio. Testing the internet connection can help determine if jitter is within an acceptable range for calls. High jitter values may cause voice packets to be delivered out of order, which can result in echo or talk-over effects. Jitter is measure in milliseconds and a good result is 5 milliseconds and a bad result is any number other than five.

• Packet loss occurs when data packets are lost during transmission over the internet. Even small amounts of packet loss can affect the quality of VoIP calls. Most VoIP tests report packet loss as an average, for example dropping 25 packets out of a thousand would be expressed as 0.25%. Sounds like a very small number and not worth worrying about; However, consider the problem if the 25 packets were dropped contiguously. At 50 packets per second that represents 50% of the time window.

No audio or static audio

• Check to make sure you turn your volume up.

• Check to make sure you plug in your headset or handset into the correct port.

• Try testing another headset/handset/speakerphone on your phone and the headset/handset on another phone to confirm if this is the problem.

• Firewall RTP ports closed.

TIPS!

• You can improve call quality by limiting other internet activities during calls.

• You should choose Ethernet connection to internet because it’s secure, has consistent speeds, and has low latency. It’s not an attractive solution—we get it. But Ethernet is just better in streaming to media centers.

• Don't put anything directly in front of your speaker. The mic is located in the front at the bottom, and this will block out the sound.

• Play around with the volume controls.

• Ensure you have a good quality headset if you use one; sometimes sound issues have nothing to do with connectivity and everything to do with a faulty headset.

• Try reducing the amount of bandwidth you are using while on a call if your Task Manager (Windows) or Activity Monitor (Mac) can show you which applications you may want to restrict before a scheduled phone call!

• Port UDP 5060 (SIP port) and UDP 10000-65000 (RTP ports) must be open on your network. Set your computer as the DMZ in your router configuration, which means that NAT essentially passes everything to you.

• Latency: less to 200ms. Check your latency here https://aws-latency-test.com/ .

• Jitter: a good value is between 0-5ms. Acceptable jitter in a call should be at or below 30ms in delay time. Higher jitter levels may negatively impact audio quality, your connection will become laggy, resulting in choppy audio.

• Bandwidth: 100 kbps is required for a single call.