Content of the material
- Prioritizing Calls
- Samsung’s upcoming foldable devices may not be as exciting as before
- Weather conditions
- NVIDIA GeForce RTX 3090 Ti is now official for $1999
- How to improve VoIP call quality
- 1) Monitor call quality
- 2) Increase bandwidth
- 3) Upgrade your router
- 4) Set up a jitter buffer
- 5) Configure QoS
- 6) Segment traffic with a VLAN
- 7) Convert to ethernet
- 8) Purchase a high-quality headset
- 9) Switch off Bluetooth
- Troubleshooting: How to improve call quality
- Hang up the speakerphone
- Choose a good headset
- Reduce bandwidth across the network
- How to Fix Packet Loss
- Advanced Voice Compression
- Understanding the Results of Your Packet Loss Test
- How to identify common call quality issues
- Packet loss
- 3. Perform a command-line ping test
The internet connection is an important factor that affects the voice quality in VOIP conversations. Bandwidth is a key factor that affects call quality. A broadband connection will work perfectly right if it is not shared with too many other communication equipment. Dependency on bandwidth is one of the main drawbacks of VOIP technology.
Another way that Ooma helps your call quality is by prioritizing phone calls so that they receive all of the network attention that they need.
In situations where you have limited bandwidth or when someone in your household is doing data-intensive internet tasks, VoIP users can experience a decline in call quality. This can cause delays in hearing what the other person in the conversation is saying, or it could be that digitized sound when a call breaks up.
While much of Ooma’s call prioritization protocols happen behind the scenes, the one key factor to remember is to plug your Ooma Telo directly into your modem, and not into a separate router.
This positions your hardware so that Ooma can intelligently prioritize your voice traffic.
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Sometimes you will feel that the voice gets distorted by static electricity generated due to natural calamities such as thunderstorms, heavy rains, floods, strong winds, strong impulses, etc. This static will not cause any disturbance when you are surfing the internet or downloading a file, but when you are listening to a voice you are surely going to complain.
It’s quite easy to get rid of this problem, all you need to do is unplug your hardware and plug it back again.
The effect of weather condition on your connection is something which is not in your control. In some cases, you can get some short-term solution, but in the majority of cases, it is in the hands of the service provider to come up with a solution. Changing the cables can solve the problem but this can be costly.
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How to improve VoIP call quality
Now that you have an idea of what’s causing your poor VoIP experience let’s dig into a number of actionable steps you can take to remedy your call quality.
1) Monitor call quality
Since there are so many reasons that your calls could be going awry, it’s valuable to monitor your call quality. Twilio, for example, offers Voice Insights to all VoIP users to track jitter, packet loss, latency, and metrics like the length of the call. By monitoring your VoIP calls, you spend less time troubleshooting call quality issues and instead proactively address any problems before it becomes detrimental to your business.
2) Increase bandwidth
If your network is not large enough to support multiple devices and users, it may be worthwhile to increase your bandwidth. Increasing your bandwidth can help accommodate a higher volume of users or devices that are connected to the network at one time.
3) Upgrade your router
Most routers for home use or for smaller businesses are pretty simple—you plug it in and don’t think about it again. But if you’re a growing company or having challenges with VoIP calls, it may be time to upgrade. Upgrading your router can give you many more capabilities, like prioritizing VoIP traffic through Quality of Service (QoS) settings, implementing a jitter buffer, and segmenting voice traffic with a VLAN. If you upgrade your router, make sure it is enabled to call via Session Initiated Protocol (SIP) and VoIP. (More on SIP here.)
4) Set up a jitter buffer
A jitter buffer helps smooth out packet distribution by collecting, storing, and sending the packets at even intervals. The jitter buffer can add a delay because of the time it takes to store and process the packets, but it can lead to a smoother experience for the end-user.
5) Configure QoS
Try configuring your Quality of Service (QoS) settings to prioritize VoIP traffic. If there is network congestion, VoIP traffic will be sent first, whereas web traffic, for example, could be deprioritized. You may have trouble quickly loading a web page, but at least your call won’t cut out.
6) Segment traffic with a VLAN
Another option to prioritize voice traffic is to set up a virtual local network (VLAN). A VLAN allows a group of devices to share a connection to specific servers even if they aren’t in the same geographical area. This can be really helpful if your company has multiple locations.
With a VLAN, you can segment voice traffic so that all voice traffic is prioritized across users. Most enterprise networks offer VLAN, so ask your network if a VLAN can be configured for your company.
7) Convert to ethernet
If you continue to have issues with an unstable internet connection, it may be time to switch from wifi to ethernet. An ethernet cable should give you a much more secure connection compared to a wireless network.
8) Purchase a high-quality headset
As simple as it sounds, a poorly designed headset can cause all sorts of shenanigans with your voice calls. From echoes to static to sound clarity, a headset can make or break your call experience.
9) Switch off Bluetooth
If you’re using a lot of Bluetooth devices, it could be affecting your connection. Wireless headphones, mouses, keyboards, etc. can all hamper your device’s internet connection, so turn off any wireless items that you’re not using.
Troubleshooting: How to improve call quality
While your VoIP provider is responsible for many things that affect call quality, there are several ways you can do a little troubleshooting and try to fix poor call quality from your end.
Hang up the speakerphone
If you aren’t having a group meeting, don’t use the speakerphone for one-on-one calls. That’s because often, a laptop microphone is of pretty low quality and picks up background noise too. Occasionally, you or your listener will experience an echo effect, especially if you have your phone and computer microphone running simultaneously.
Take calls on a good pair of headphones or take the call on your mobile or desk phone.
Choose a good headset
If you’ve ever been in an open office setting, you know how distracting noise can be. The right headset can mitigate or eliminate much of the noise, and they often come with excellent microphones. Some headsets have noise-canceling features built in that include dual-mode active noise cancellation (ANC) that adjusts to your environment.
Wired headsets, either corded or USB, often deliver a clearer conversation than Bluetooth or other wireless models.
Reduce bandwidth across the network
Network congestion is a primary cause of jitter. When there’s too much traffic across the network, collisions become inevitable, rather like a highway at rush hour.
Your IT department can use most network monitoring tools to identify the packet streams carrying VoIP data on the network. Once identified, this data can be tagged for protection using Quality of Service (QoS) features in the router.
They can manage bandwidth by measuring network activity and adding hardware and software to take some of the load. You can also adjust your router settings to, say, prioritize your VoIP calls instead of your kids video game streaming. (If you need a single, particularly robust line for video conferencing, you may be able to set up a dedicated virtual network just for VoIP traffic.)
How to Fix Packet Loss
Packet loss doesn’t have to be an issue that causes you daily headaches. By taking a few steps upfront, continually monitoring your system for high quality, and getting professional help when you need it, your employees and customers will be able to focus on their conversations without disruptions and distractions.
Poor voice calling quality can create big problems for your company. Packet loss not only disrupts voice calls, but it can also slow down your file transfers and cause problems with data security. In the end, it can damage your brand’s reputation.
Here are some steps you can take to fix the problem of packet loss:
- Ensure your hardware is always up to date. Having the most recent updates on your software programs will ensure your cloud phone system is working at an optimal level. The same should be said for any phone applications that your teams are using.
- Bandwidth matters. Make sure your office and your teams, whether remote or distributed, have the proper amount of bandwidth.
- Make sure all of your internet cables are in good working condition. Also make sure that you put devices in a safe location that will not be disrupted.
- Reduce radio signals and any other signals that could be interfering with your Wi-Fi connection.
Suppose you continue to experience problems with poor phone call quality, latency, jitter, or data loss. In that case, you might consider getting a professional involved to help clear up the problem and get tips about avoiding it in the future.
Advanced Voice Compression
All VoIP phone service providers transmit your phone calls over the Internet. And because there are limits to your Internet speeds, the less data is required for audio, the more reliable are your phone calls. And this efficiency contributes to increased clarity.
Ooma’s advanced voice compression algorithm reduces bandwidth consumption by 60% over standard VoIP technology. Additionally, the way that the voice data is compressed makes it more capable of withstanding packet loss without degradation.
Understanding the Results of Your Packet Loss Test
As you gather your packet loss test results, let’s take a closer look at some of the terms and measurements you need to know.
Latency (or Ping) — This term refers to the delay between a user’s action and the web’s response to it. In simpler terms, it refers to the amount of time it takes for data packets to make the entire trip.
Jitter — This term refers to the variation in the times that various data packets arrive. The longer it takes for all the packets to arrive, the more likely calling parties will experience audio problems.
Data loss — This term refers to a system error where information gets lost or destroyed during storage, transmission, or processing.
How do you define good call quality?
For VoIP calls, you should aim for under 150ms ping and 30ms jitter for optimal call quality.
Latency, jitter, and data loss aren’t much of a concern if you’re browsing around the internet or checking in on your social media accounts. For business purposes, these terms have a lot more significance because they impact call quality. Phone communication is essential for small businesses, and you’ll want to be sure the call quality is consistently good. Poor call quality will create a bad customer experience, which most likely will lead to customer churn.
How to identify common call quality issues
Three common issues can wreak havoc on clear communication using VoIP: packet loss, jitter, and latency:
VoIP breaks sound into tiny digital packets that move along the network. They don’t all necessarily move in order or even along the same route. The goal is to get those packets to their endpoint as quickly as possible. Once there, they are reassembled in the correct order for natural sound to issue from the speaker.
But what if some of the packets are lost? If there are network issues, packets may be dropped inadvertently on their way to the endpoint. Missing packets result in sound loss experienced by callers as a drop in audio.
If the packet loss is severe enough, the call is partially or even entirely dropped.
You may have heard jitter during a call. It occurs when there is a change in the amount of time it takes for the packets to travel across the network. They get stuck somewhere, often because the network is congested.
The result is sound that arrives at the caller at irregular intervals, which creates a choppy audio sound.
Latency is somewhat related to jitter. It has to do with the amount of time it takes a packet to travel across the network. Instead of irregular intervals, the entire message arrives later than intended. Callers experience “double talk,” even if they stop speaking ahead of hearing audio from the other end.
The audio delays make conversation tricky, with people accidentally speaking over each other.
With these three common issues in mind, how can you measure VoIP (HD) call quality?
3. Perform a command-line ping testBefore you begin: For the most accurate results, run this test when internet usage is high on a computer that is hard-wired (not connected through Wi-Fi) to the same network as your phones.
- Perform a ping test:
Option Description Mac
Open Terminal and then run:
Open Command Prompt and then run:
- Compare your results to the minimum requirements.
Result Requirement Packet Loss <0.5% Ping <100ms Jitter Gaps >10ms between the minimum and average ping times.
- If your results are inconsistent or insufficient:
What to do next: If your results are not showing any issues with these metrics and your problem persists, contact Customer Support for further help troubleshooting.