Identifying common call quality issues: Difference between revisions
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*VoIP requires a certain amount of bandwidth to transmit voice data without interruption. Each concurrent call needs to have 80 kbit/s of bandwidth available (both upstream and downstream). | *VoIP requires a certain amount of bandwidth to transmit voice data without interruption. Each concurrent call needs to have 80 kbit/s of bandwidth available (both upstream and downstream). | ||
*Latency, or the delay in transmitting data over the internet, can affect the quality of VoIP calls. Testing the internet connection can help determine if latency is within an acceptable range to make and receive VoIP calls. | *Latency, or the delay in transmitting data over the internet, can affect the quality of VoIP calls. Testing the internet connection can help determine if latency is within an acceptable range to make and receive VoIP 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 VoIP calls. To measure the variation over time of latency across the network, you need a jitter test. 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. | |||
*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 VoIP calls. To measure the variation over time of latency across the network, you need a jitter test. 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. Testing the internet connection can help determine if packet loss is within an acceptable range for 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. | *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. Testing the internet connection can help determine if packet loss is within an acceptable range for 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. | ||
Revision as of 14:33, 23 April 2024
When it comes to call quality, there are many variables and factors that can cause disruption.
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 80 kbit/s of bandwidth available (both upstream and downstream).
- Latency, or the delay in transmitting data over the internet, can affect the quality of VoIP calls. Testing the internet connection can help determine if latency is within an acceptable range to make and receive VoIP 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 VoIP calls. To measure the variation over time of latency across the network, you need a jitter test. 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. Testing the internet connection can help determine if packet loss is within an acceptable range for 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!
- Don't put anything directly in front of your phone if you plan on using the 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!
If your problem persist
If you are unable to get the configuration working, please contact us at itis.cmsupport@softtek.com. To help us resolve this quickly, please include:
- -Your Public IPv4: (Found at https://checkip.amazonaws.com/)
- -Screenshot of Settings:
Ctrl+P
- -Screenshot of Account:
Ctrl+M
- -Issue Description: A brief summary of the problem you are facing.