Your browser does not support JavaScript!

What is Hosted VoIP?

Are you looking for a phone system that offers affordability, flexibility, modern capabilities, and so on. Or perhaps you’re planning on modernizing your current communication setup.

Either way, the solution is simple – hosted VoIP.

In this article, we’ll define hosted VoIP, how it works, its benefits, key business features, and use cases.

Understanding Hosted VoIP and How it Works

Before delving into hosted VoIP, you must first understand VoIP on its own. Voice over internet protocol (VoIP) digitally transmits voice calls over an IP network. This allows you to make and receive business calls through the internet.

To accomplish this, it uses packet switching technology, which converts your voice data into digital packets. These packets are then sent to the receiver via the internet. And once they arrive at the destination, the packets reassemble into voice. This process occurs in a matter of seconds and allows callers to communicate seamlessly.

Now, this brings us to hosted VoIP. There are two general ways to deploy VoIP: hosted or on-premise.

With hosted VoIP, a third-party provider “hosts” and manages the infrastructure on their premises. They then supply cloud communication services to your business through a network connection.

This deployment model allows you to access hosted VoIP services from any IP-enabled device or location. Additionally, this setup is far more affordable than on-premise solutions. This is because you don’t have to worry about routine maintenance, monitoring, and upgrades.

With an on-premise model, your business owns and manages all VoIP infrastructure. In other words, the equipment typically resides on-site. While this option offers full control, it is expensive to set up and maintain as it requires plenty of free space, energy, monitoring, and upkeep.

How Can Your Business Benefit from VoIP?

Many companies choose VoIP as their business phone solution because of its many advantages and capabilities. Here’s how your business can benefit from hosted VoIP services:

• Substantial cost-savings – Save on additional equipment and routine maintenance while avoiding costly international and long-distance calling fees.

• Increased flexibility and mobility – Accessible from any location or device, providing more mobility than traditional deskphones.

• Highly scalable – Upgrade or downgrade your service when needed without changing your entire system.

• Enhanced call quality – Experience clearer sound and better call quality, since VoIP calls travel faster than over traditional phone lines.

• Easy configuration – Integrate it within your existing system, as no additional equipment is required.

• Improved functionality – Enhance your business phone system’s functionality with features only accessible through the cloud.

An image showing the ways a business can benefit from using hosted VoIP.

Hosted VoIP Key Features

As mentioned above, many VoIP providers offer users a number of advanced cloud features with their services. This way, you can effectively manage and improve your business’ communication. Let’s take a look at the top hosted VoIP features:

  • Phone numbers (local, international toll-free, etc.)
  • Call recording
  • Caller ID management
  • Advanced IVR
  • Call flow designers
  • Call routing and forwarding
  • Failover capabilities
  • Softphones or mobile apps
  • Voicemail to email
  • Call detail records and analytics
  • Integrations / APIs

4 VoIP Use Cases

With a clear picture of hosted VoIP and its capabilities, let’s discuss how companies can use this technology to grow their business.

1. Expand Globally

IP telephony is not tied to a specific location. So, you can use it to expand globally, enter new markets, and access a broader customer base. The best part? You can do this without opening a physical location or increasing overhead costs. Simply set up cloud phone numbers in your desired countries or markets. Then, forward incoming calls to your business headquarters.

2. Improve Network Reliability

With access to hosted VoIP’s failover strategies, you can build your network’s redundancy by minimizing downtime and preparing for potential outages. So if your system experiences an interruption, calls are automatically rerouted to an alternate, predetermined location.

3. Offer 24/7 Global Support

Cloud communication services are typically fully customizable to your business’ unique needs. This includes predetermining routing rules and features like time-based or location-based routing. You can use them to offer customers 24/7 global support, increasing customer satisfaction and accessibility.

4. Connect Distributed Teams

You can manage hosted VoIP services in one centralized location and from any location or device. This means teams can communicate via smartphones, computers, tablets, and desk phones. These capabilities make it the perfect solution for connecting local and distributed teams.

Get Started with Hosted VoIP

As you can see, hosted VoIP helps keep costs down, improve business communications, enhance call quality, and much more. All you need to get started:

  • a stable internet connection with adequate bandwidth,
  • an IP-enabled device or media gateways,
  • and a reliable provider.

United World Telecom provides enterprise-grade VoIP phone services for businesses around the world. After 26 years of telecom experience, we’ve established long-term relationships with reliable Tier-1 carriers across the globe. This enables us to deliver users with high-quality voice services.

Reach out to learn more about our reliable service and if we are the right VoIP provider for you. Speak with our dedicated telecom experts at +1 (561) 276-7156 or chat with us online today!

VoIP versus UCaaS: Understanding the Difference

More and more SMBs are switching to cloud communication solutions and moving away from landline systems. But why?

Companies adopting hybrid and fully remote workforces need communication solutions that stretch across geographic locations.

This opens up the space for more advanced communication technology such as VoIP, UCaaS, CCaaS, CPaaS, and more. So, how do you decide what solution is best for your business?

This article will look specifically at VoIP versus UCaaS, so you can better decide what phone system will improve your team’s productivity.

Building a Business Phone System: VoIP versus UCaaS

As a network professional, you’ve certainly heard about VoIP before. Voice over internet protocol and IP telephony solutions replace traditional PSTN by using the internet to route calls.

On the other hand, unified communications as a service (also known as UCaaS or UC) groups together various communications technologies into one centralized platform.

You’ll notice in the comparison below that both VoIP and UCaaS have very similar benefits. However, the main difference lies in how they are built and how you can use them.

One way to understand the difference between these services is that VoIP makes UCaaS solutions possible.

VoIP, for example, refers to one specific communication technology, whereas UC is an umbrella term for many different types of communication technologies. Therefore, it is not an easy, 1-1 comparison.

VoIP is typically a voice service that handles inbound and outbound calls. UCaaS, in comparison, brings all communication channels (voice, video, messaging, chats, etc.) into one platform and enables them using IP.

Because of this key difference, VoIP is less encompassing than Unified Communications.

Let’s look at these solutions more closely.

A comparison of VoIP versus UCaaS.

What is VoIP and How Does it Work?

VoIP was designed to provide a landline’s functionality along with cloud-based service’s flexibility and capability. This enables phone calls through an internet connection, removing location-related restrictions.

And since VoIP is software-based, you can easily integrate it with other similar cloud communication software and build a wholesome phone system.

What can you do with VoIP?

  • Place and receive calls online
  • Forward calls from a desk phone to a mobile phone
  • Access call recordings and logs
  • Utilize voicemail to email forwarding
  • Access advanced calling features such as IVR, caller ID management, and more
  • Automate call routing based on various rules
  • Easily manage dial-in conference calls.

Benefits of VoIP:

  • More affordable and helps many businesses drastically reduce costs
  • Easier to scale, both locally and globally
  • Supports local, remote, and global teams through one phone system
  • Provides uninterrupted, high voice quality.

What is UCaaS and How Does it Work?

UCaaS solutions go a step further than VoIP by bringing more than voice to your phone system. As mentioned above, it gives you access to multiple communication channels and apps under one cloud-based roof.

This way, your teams can communicate with each other and customers through the channels they prefer.

And by streamlining communication through one platform, your teams can be more productive as everything they need is located in one place.

What can you do with UCaaS?

  • Integrate email, SMS, video conferencing, chat apps, etc., into your phone system
  • Access various communication features and apps
  • Centralize call management from one platform.

Benefits of UCaaS:

  • Cost-efficient and scalable solution
  • Use and manage from anywhere in the world
  • Supportive of global and distributed workforces
  • No hardware or maintenance required.

Related: 5 Unified Communications Trends You Need to Know in 2022

Which Communication Solution Should I Choose?

As you can see, there are benefits to both VoIP and UCaaS solutions.

However, which solution you need depends on your requirements and resources.

For instance, enterprise-level businesses might find more use in a UC solution that supports their teams and provides more communication and collaboration options. However, if you’re an SMB that needs only voice support across local and remote teams, then a VoIP solution will meet your needs.

One important thing to note is that both solutions will allow you to centralize and scale your solutions as needed. VoIP lets you consolidate different local, regional, and global voice carriers under one carrier (i.e. your VoIP provider). And UCaaS lets you consolidate different communication channels and apps under one service provider.

In fact, you can even add a VoIP service to your UC stack for voice support. By consolidating, you reduce your overall total cost of ownership (TCO) across different channels or providers.

So, whether you start with VoIP or UCaaS, you can grow your communication stack at your own pace via integrations and APIs. The real question is — what do you need right now, and what can support your plans for the future?

Learn how to scale with VoIP by speaking with one of our telecom experts today! Call us at +1 (561) 276-7156 or chat with us online!

What is Automatic Number Identification (ANI)?

In this post, we’ll go over the ANI telephony feature and how businesses use this feature to provide better support and sales.

What Does ANI Mean in Telephony?

In telecommunications, ANI refers to Automatic Number Identification. This phone feature, offered by many cloud telephony and VoIP providers, works in conjunction with call data and is used primarily for billing purposes. Let’s find out how:

What is Automatic Number Identification?

Automatic Number Identification allows the recipient of an incoming call to determine and display the number of the phone that originated the call. In other words, it displays the number of the person dialing or placing the call.

Before ANI, telephone operators manually requested the phone number of the person calling, especially for a toll call. But now, telecom providers can use this service to help users (your business) understand and analyze their call data, volume, and traffic.

Are ANI and Caller ID Services the Same?

This feature is often understood in relation to caller ID services; however, they’re not the same since they utilize different underlying technology.

Automatic Number Identification is the Billing Telephone Number (BTN) used by carriers and assigned by the telco switch. Caller ID, on the other hand, is the display number provided by the caller’s equipment (VoIP or PBX setup) or originating carrier (outbound calling service).

Users can change the outgoing caller ID when making outbound calls. However, one cannot change or block the ANI number.

An image of automatic number identification for international calling.

How is ANI Used in Businesses and Call Centers?

1. Use call detail records for billing purposes:

Automatic Number Identification is especially useful in cases where incoming calls (toll and toll-free) are charged based on the caller’s (your customer’s) number and location. For example, toll-free costs for incoming calls vary depending on where your customer is calling from or where the call originated from.

By identifying where your calls come from, you can better understand your VoIP phone bill. This includes:

  • local, regional, and international call charges
  • how much you pay as a customer
  • and, sometimes, what your VoIP provider owes its own carriers.

ANI systems and reports keep track of your call data. And in some advanced reports, you can even see specific charges for each call. Based on this information, you can then make adjustments to your phone service accordingly.

2. Use call data to improve customer experience:

Such call data can also give you key insights into where your calls come from and, by extension, where your customers are located. Once you’ve analyzed this call data, you can decide how to serve them best.

For instance, say you have a lot of international clientele. With advanced call routing, businesses and call centers can route calls based on a variety of preset rules. So, you can forward calls from specific regions to a call center or support team closer to the destination — all based on the caller ID and area code of the caller.

Or, if you have enough after-hours call traffic, you can outsource those calls to a remote agent or your personal phone. Such time-of-day routing lets you offer 24/7 support.

How United World Telecom Can Help

United World Telecom uses ANI in our system for billing purposes. This enables our customers to see the caller ID of their callers. And our call detail records and billing reports can help you understand call traffic and identify new growth opportunities. Additionally, they also have access to features like location-based routing so they can serve customers wherever they are located.

To learn more, call us today at 1 (877) 898 8646 or chat with us online!

What is G.711? And Why is this VoIP Codec Important?

When setting up a new VoIP phone system, you must equip the right voice codec to ensure reliable and clear call quality. The type of codec you need depends on your provider and phone system.

In this article, we’ll go over the G.711 codec and why it is the preferred codec for VoIP calling.

VoIP Codecs and Why They are Important

VoIP codecs are designed to convert analog voice signals to digital packets (compression) and then reassembled back into audio (decompression) when they arrive at their destination.

In doing so, they establish and maintain VoIP call quality and determine bandwidth use for incoming and outgoing calls. So, to ensure your teams can communicate effectively through your VoIP phone system, you will need to use the right codec — one supported by your provider.

You can adjust these voice codecs depending on what you need, such as better quality, evenly distributed bandwidth, etc.

What is G.711?

G.711 is a commonly used VoIP codec that converts voice signals to digital packets with an output of 64kbit/s.

This codec uses packet loss concealment (PLC) to minimize the effect and impact of packets dropped during transmission. It was established in 1972 as the default pulse code modulation or PCM standard for IP PBX and PSTN networks.

There are two main algorithms for the G.711: The μ-law codec used in North America and Japan and the A-law codec used in the rest of the world. Additionally, G.711 μ-law offers more resolution to higher range signals. And the G.711 A-law offers more quantization at lower signal levels.

Why Choose the G.711 Codec for VoIP?

While there are a few different codecs for VoIP, the G.711 is the most preferred and most commonly offered by VoIP providers. And there’s a good reason for that. Here are a few reasons why the G.711 is one of the best codecs for VoIP calling:

  • Two variants for worldwide usage (μ-law and A-law)
  • High MOS call quality score of 4.2
  • Uncompressed high-quality voice
  • High bandwidth requirement
  • Focus on precise speech transmission
  • Good for LAN and VoIP to PSTN setups
  • Most reliable call quality

The other popular voice codec is the G.729 codec, which uses less bandwidth since it compresses packets. But this also means it sacrifices quality due to compression.

Why Do You Need to Care About Bandwidth for VoIP?

Since VoIP calls travel over the internet, they require a certain amount of bandwidth to efficiently transmit voice data packets back and forth. This means your VoIP calls will compete with other internet traffic. And if you don’t have enough bandwidth or wisely distribute traffic across your network, your call quality will be affected.

So, how much bandwidth do you need? This depends on how many calls you expect to run concurrently and what else your teams use the internet for.

Consider these numbers:

Number of Concurrent Calls Bandwidth Recommended
1 100 Kbps
5 500 Kbps
10 1 Mbps
15 1.5 Mbps
20 20 Mbps

Typically, the bandwidth needed for each concurrent VoIP call resides anywhere from 85-100Kbps. And the G.711 codec consumes 87.2kbps of bandwidth.

Codec Bitrate Bandwidth Usage
G.711 64 Kbps 87.2 Kbps
G.722 48-64 Kbps 80 Kbps
G.723.1 5.3 Kbps 20.8 Kbps
G.726 32 Kbps 55.2 Kbps
G.728 16 Kbps 32 Kbps
G.729 8 Kbps 31.2 Kbps

So, if you have an internet connection of 500kbps, you can theoretically run at least 5 calls simultaneously.

VoIP Codec Supported by United World Telecom

United World Telecom supports the G.711 codec (both μ-law and A-law), and our call quality has an average MOS of 4.3. We can help you set up a VoIP phone system that works best for your communication needs, guaranteeing high call quality and network reliability.

To learn more, call us today at 1 (877) 898 8646 or chat with us online!

Is VoIP Reliable?

Tons of businesses are switching to VoIP for its numerous benefits — from affordability to flexibility. But many new users wonder if VoIP is dependable for business calls, especially since internet-based calling earned a bad reputation in the past.

In this article, we will answer commonly asked questions about voice over IP solutions, including — is VoIP reliable?

Reliability of VoIP

Voice over IP (VoIP) is an internet protocol that converts audio into digital packets and then transmits these packets to the destination. By doing so, VoIP enables users to make and receive calls using the internet.

But many users worry that this is not a dependable solution. This fear stems from a time when these internet-based tools weren’t developed enough, resulting in quality issues.

So, is VoIP reliable? Yes, VoIP can be very reliable, despite concerns surrounding cloud-based calling.

In fact, with how cloud calling has evolved over the years, VoIP phone systems are often preferred over traditional POTS because they offer crystal-clear call quality and high uptime.

But, to enjoy this, you need to take the proper steps to minimize downtime and keep your system running efficiently. This often involves both the provider and the user. Let’s look at the client-side and provider-side factors.

Client-Side Factors

So, what can you do as a user to ensure your VoIP phone system runs smoothly? Here are some key factors to consider:

Bandwidth requirements:

VoIP requires adequate bandwidth to ensure calls travel uninterrupted. This means you need a stable, high-speed internet connection with enough bandwidth dedicated to voice calls. When deciding how much bandwidth you’ll need, consider the following:

  • Number of concurrent calls
  • Number of phone lines
  • Other applications working simultaneously
  • Codecs supported by your provider, etc.

Router optimization:

Similarly, you want to make sure your router is optimized as well. This means investing in a high-quality internet connection, whether WiFi or wired connections. It also means adjusting your router’s settings to prioritize VoIP calling (and voice packets) over other traffic (such as streaming and browsing). You can use quality of service or VoIP QoS to adjust traffic priority.

Backup power and internet sources:

As part of your disaster recovery plan, you will also need backup services in place to get your system up and running during outages or other disasters. Plan to have a backup mobile connection and power sources.


Finally, you must make sure you have the right equipment to make and receive VoIP calls effectively. This covers everything from routers to softphones and headsets.

In most cases, your VoIP provider can make recommendations. But it is a good idea to invest in high-quality, echo-canceling headsets for your agents and use softphones offered by your provider to make calls from your devices. On top of that, ask your provider about the router and internet settings needed to support your VoIP calling needs.

Provider-Side Factors

You also want to ensure your provider is prepared to offer you quality service with a trustworthy support team.

So, what makes a VoIP provider reliable?

99.999% uptime:

Your provider should supply you with more information that determines the quality of the service your provider will deliver. This must-have information will outline the expected uptime, call quality, troubleshooting assistance, and other essential information pertaining to your service.

Global and local servers:

Most VoIP providers servicing enterprise-level organizations usually have their own VoIP infrastructure with points of presence (PoPs) spread across the world. This ensures calls are distributed reliably, even if one route fails. But some providers work with a vast network of local and regional operators to bring you the same level of service reliability. When choosing a VoIP provider, ask how they route their local and global calls and if any areas are not covered within their network.

Dedicated account management:

Another key factor that determines your VoIP service quality is having a dedicated account manager who understands your business communication needs well and works to ensure you get the best out of your service. Choose a provider who offers an account manager for no extra charge. This way, you know your service is in good hands.

24/7 technical support:

Similarly, you also want a provider with a responsive tech support team in case of emergency situations. Look for a provider with multiple support options such as live chat, 24/7 phone support, email, trouble tickets, troubleshooting guides, etc.

Regular monitoring:

Additionally, your provider should constantly monitor their networks and servers. This is the best way to ensure calls travel accurately and quality remains high. Most providers use a quality monitoring service or software to keep tabs on their voice network. Ask your provider how they monitor quality and how they can guarantee uninterrupted service.

Is VoIP More Reliable than POTS?

Most VoIP newcomers wonder how exactly VoIP is more reliable than POTS. But VoIP is statistically a better choice than PSTN for small to medium-sized businesses.

To better understand why businesses prefer VoIP over landline systems, you must look at their differences.

Here’s a brief overview of PSTN versus VoIP:

  • VoIP is more cost-effective than PSTN. In fact, cloud telephony providers can offer better quality and more features at a lower cost than traditional landlines.
  • You have access to advanced voice calling features with VoIP. PSTN is relatively limited.
  • VoIP use has steadily increased while landline use has steadily decreased over the past two decades – especially in the US.
  • Internet-based calling allows users to use any IP-enabled device to place calls, adding geographic mobility.
  • With redundancy and failover capabilities, you can ensure your VoIP system is up and running, even during a power or internet outage.
  • VoIP infrastructure continues to evolve while PSTN is being phased out.
  • And finally, VoIP offers more flexibility, growth, and scalability than a PSTN setup.

VoIP reliability statistics.

Source: Statista

So, the question remains, do you want to base your communication system on a technology that may not be as functional or reliable 10 years from now?

Source: Bullseye Telecom

When is VoIP the Better Choice?

VoIP can help you increase productivity, reduce costs, communicate effectively, and run your business from anywhere. But how do you decide if your organization can benefit from this switch?

According to our Sales Manager, Luke Genoyer, here is when VoIP is the right choice for a business:

  • Young companies who want to set up a reliable and easy-to-manage phone system can benefit from month-to-month services with no obligation as well as a fast, inexpensive, and easy setup.
  • For businesses that need to scale or plan to grow — in size or global reach — in the next few years, VoIP makes it easy to add and remove users and numbers easily.
  • Businesses that need access to call center data to analyze their call traffic and build a more responsive support team to improve customer experience.
  • Use international calling to establish a local presence around the world with various local and international phone numbers.

How to Choose a Reliable VoIP Service Provider

While there are many VoIP and cloud telephony providers on the market, you want to find one that meets your communication needs and budget. Finding this provider will take some effort and research on your part. Here’s a quick checklist of things to consider when looking for a new VoIP service:

  1. Determine communication needs and budget.
  2. Evaluate features and pricing.
  3. Review inventory and global coverage.
  4. Check uptime and carrier reliability.
  5. Read reviews and customer testimonials.
  6. Try for free, if available.

Get VoIP with United World Telecom

United World Telecom has offered VoIP services to businesses for over 25 years. Our experience in the telecom industry has resulted in strong relationships with Tier-1 carriers around the world. This means we can provide businesses of all sizes with high-quality and reliable VoIP phone services.

Want to learn more about our cloud telephony services? Reach out to our telecom experts and chat with us today!

How to Troubleshoot One-Way Audio on VoIP Calls

Whether you’re a VoIP user or not, you’ve probably experienced one-way audio during a phone call. While VoIP offers businesses many benefits and modern capabilities, it’s normal to experience audio issues periodically. So, in this article, we’ll go over how to troubleshoot one-way audio. This way, you can solve this issue as quickly as it appears.

What is One-Way Audio?

One-way audio happens when the receiver can’t hear you, but you can hear them (or vice versa). It’s a common VoIP issue that impacts your ability to communicate during a call. And one-way audio often leads to a constant exchange of “Hello?” and “Can you hear me now?” – which is frustrating for both parties on the line.

A diagram showing one-way audio on VoIP calls.

Why Does It Occur?

It’s possible for one-way audio to occur during the beginning or middle of an existing call. Disruptions to the audio stream cause this issue and appear for a number of reasons.

The most common causes include:

  • Equipment issues
  • Firewalls / NAT blocking voice data
  • Incompatible codecs
  • Routing misconfiguration

We will look at each of these one-way audio causes and explain how to troubleshoot them.

How to Troubleshoot One-Way Audio: 4 Tips

Although one-way audio is a typical issue among VoIP calls, it is usually easy to diagnose and fix. Let’s look at how to troubleshoot one-way audio.

1. Check Your Equipment

Faulty equipment leads to many VoIP call quality issues, including one-way audio. So, the best way to start troubleshooting one-way audio is to rule out your equipment as the cause.

Equipment Issue Solution

Verify that all your equipment (headset, microphone, and desk phones) is properly connected by checking all cables and selected inputs on your devices. Then, inspect your hardware for any visible damage.

Once you’ve assessed your equipment’s physical condition, test your line. For all setups (softphones, IP phones, and legacy phones), simply initiate a test call and see if you can hear audio on both ends. If you’re using a softphone, you can also use any audio recording software to ensure your inputs are correctly configured.

2. Review Your Router’s Settings

Voice over IP uses packet switching technology to deliver calls through the internet. So, in order for these calls to succeed, voice data must travel from one point to another. This means these data packets must pass through your network’s firewalls – which can block audio packets from transmitting to their destination.

It’s worth noting that Network Address Translation (NAT) acts as a firewall and may also be causing one-way audio.

Firewalls / NAT Blocking Voice Data Solution

To troubleshoot this type of one-way audio issue, make sure the firewall ports are open to your provider’s recommended settings. For United World Telecom users, open ports 10,000 to 60,000. Additionally, whitelist your provider’s IP address in your firewall’s settings.

If that doesn’t work, the problem may lie with NAT. First, evaluate your network. Do you have more than one router supplying NAT? If so, you may be double NATing your traffic – leading to data arriving at the wrong destination. The best way to fix this is to turn off the extra instances of NAT.

After that, if you’re still experiencing one-way audio, try equipping SIP ALG. This network component allows your call data to pass through your firewall’s security checks and NAT rules. However, it’s critical to note that if SIP ALG is implemented incorrectly, it will lead to other problems, which is why most providers recommend leaving it off.

Depending on your setup, you may need to work with your provider to understand the best method of dealing with NAT or SIP ALG-related issues.

3. Ensure Devices Share Common Codecs

When you place a VoIP call, the two endpoints communicate and select a VoIP codec available to both devices. But if this process is unsuccessful, callers may experience one-way audio – as voice packets cannot be appropriately exchanged.

Incompatible Codec Solution

Make sure both endpoints support a common codec and that your line is set to the proper codec. Contact your provider with questions about correct codec names. For our users, use codec G.711 for all lines and SIP trunks. The United World Telecom network supports the G.711 codec, which provides the best VoIP call quality and uses no compression.

4. Reconfigure Routes

Network routing technology directs your call data from its source to its final destination. While your audio may successfully arrive at the endpoint, this doesn’t guarantee that you’ll always receive voice data in return. For example, If the call path from point A to B has a low-latency connection, but the data from point B to A takes a different path with high latency – you may experience one-way audio. These are both indications that routing misconfiguration may be present.

Routing Misconfiguration Solution

Many telecom professionals suggest troubleshooting this one-way audio issue using a “bottom-up” approach. This approach requires you to assess your IT infrastructure from the physical layer to software components to pinpoint where the misconfiguration is.

If you’re operating on a hosted VoIP service, this issue may lie on your provider’s side. So, work with your VoIP provider to diagnose the misconfiguration and fix the problem.

Solve Call Quality Issues with a Reliable Provider

With the right provider, you can avoid call quality issues altogether. And if they do come up, it’s vital that you solve them quickly. Look for a VoIP provider that understands your business communication needs and offers reliable services.

United World Telecom delivers high-quality hosted communications. We offer 24/7 support and a dedicated account manager for every user. This dedication helps avoid VoIP problems before they begin.

For more troubleshooting help or to learn more about our services, call us at 1 (877) 898 8646 or chat with us online!

What is VLAN? Understanding Virtual Local Area Networks

Here’s a brief overview of how virtual local area networks work and why businesses need this type of network. Learn about the different types of VLAN and what options are available for those interested in VoIP.

What is a Virtual Local Area Network?

A Virtual Local Area Network (VLAN) is a local computer network that breaks up a single switched network into a set of overlaid virtual networks. It groups together subsets that share one LAN while separating network traffic for each group.

The virtual LAN breaks up network traffic so that only the devices within the necessary virtual network receive it. This helps your IT infrastructure avoid network congestion.

A diagram depicting a VLAN flow chart.

What Do You Need to Set Up Virtual Local Networks?

With virtual local area networks, you don’t usually have to buy additional equipment. You can set up a VLAN with most existing switches, access points, and routers.

Virtual LANs operate at the link layer, Layer 2 switch. And they can be used as a single physical link or extended to multiple links by additional Layer 2 switches. You will also need Layer 3 switches for each individual virtual LAN. And you will use Ethernet cables to connect all these components to the router. Depending on your setup, you may even benefit from VLAN trunk links that carry more than one VLAN.

What Does a VLAN Do?

Medium-sized businesses and large enterprises use VLANs to partition, divide, and manage network traffic efficiently. For instance, a company can separate its sales department traffic from its human resources traffic by assigning different virtual networks for each.

Similarly, you can even arrange all voice traffic from deskphones or IP phones to fall within one network and other devices on another network. This allows you to maintain high voice call quality with little-to-no delay or lags.

This way, you reduce network congestion by efficiently managing and distributing network traffic. And this further helps improve network performance on a much larger scale.

A comparison of LAN vs WAN vs VLAN.

Types of VLAN

Now, there are 5 main types of virtual local area networks. Here’s a quick overview of each type:

Default VLAN

Refers to the one network that a device’s ports belong to when switched on – especially with a new switch. The default is VLAN 1 for most switches. It is important to change this for security; however, note that you cannot rename or delete this virtual LAN.


Divides the network into a group of users and a group of devices. Also known as user VLAN, this is only used for carrying user-generated data.

Voice VLAN

Configured to carry voice traffic, voice VLANs usually have high transmission priority over other types of traffic. This separate virtual local area network ensures VoIP call quality by preserving bandwidth reasonably.

Management VLAN

Equipped to manage infrastructure and access management capabilities of a switch like system monitoring, logging, and so on. This virtual local area network is also used to separate management traffic from other traffic and saves enough bandwidth for management even when general user traffic is high.

Native VLAN

Allocated to an 802.1Q trunk port, which places untagged traffic (traffic not tagged by any virtual local area network) within the native VLAN.

Using VoIP with VLAN

Since virtual LANs segment and separate traffic, VoIP users can benefit from separating data and voice traffic within one network. You can apply VoIP QoS or Quality of Service settings via VLAN tagging to provide special high priority to voice traffic.

Such segmentation also prevents VoIP devices from competing with traffic from other devices. This reduces overall delays and improves VoIP call quality. And you can quickly troubleshoot VoIP issues since that traffic is separated with virtual LANs.

Want to learn more about using VoIP with VLAN? Our tech experts at United World Telecom can help you identify the best VoIP solution for your business. Call us today or chat with us online to learn more! 

Guide to VoIP Codecs and How They Affect Call Quality

Voice over IP (VoIP) allows businesses to communicate with customers near and far with reliable voice quality through the internet. To understand how to get the best voice call quality from your VoIP phone system, you need to pay attention to voice codecs.

So, how do VoIP codecs support VoIP call quality?

In this guide, we will go over:

  • What are VoIP Codecs?
  • What Voice Codecs are used for VoIP?
  • What is the Best Codec for VoIP?

Let’s dive in!

What is a VoIP Codec?

A VoIP codec is a technology that establishes the audio quality, bandwidth, and compression of VoIP calls. The term codec is a portmanteau of Compression and Decompression.

When placing calls with VoIP, the voice needs to be encoded and converted into data packets. During this process, data is compressed to increase transmission speed and improve caller experience with crystal-clear voice.

This is where codecs come in, as they help encode and decode voice.

Why are Codecs Important for Voice over IP?

VoIP codecs convert analog voice signals into digital data packets (compression) and then convert them back to voice at the destination (decompression).

Because of this crucial process, these codecs determine the quality of your VoIP calls. Specifically, they influence latency, packet loss, and other VoIP call issues that may occur when calls travel over the internet.

Users can adjust these voice codecs to meet different needs, such as improving voice quality or reducing bandwidth consumption. You can work with your VoIP provider to understand what codecs they use for their service and how that may impact your communication.

What Codecs are used for VoIP?

Here’s a list of common VoIP codecs:

Codec Bandwidth
(kbit/s – bit rate)
Key Points
G.711 64 kbit/s
  • Focuses on precise speech transmission
  • Two variants: μ-law (US and Japan) and A-law (Europe)
  • 8 kHz sampling frequency
  • Compression ratio 1:2 – 16-bit samples into 8 bits
  • Requires high bandwidth
  • Good for LANs
  • High MOS of 4.2 when conditions are met
  • No licensing fees
  • Best codec for VoIP-PSTN connections
G.722 48 kbit/s
56 kbit/s
64 kbit/s
  • High-definition voice codec
  • 16 kHz sampling frequency
  • Adapts to varying compressions
  • Improves audio quality
  • Lowers latency
  • Better quality and clarity
  • Free
G.723.1 5.3 kbit/s
6.3 kbit/s
  • High compression
  • High-quality audio
  • Low bandwidth requirement
  • Works with dial-up
  • Requires more processor power
G.726 16 kbit/s
24 kbit/s
32 kbit/s
40 kbit/s
  • 8kHz sampling frequency
  • Most used mode – 32 kbit/s
  • Commonly used on international phone trunks
  • Standard codec for DECT wireless phone systems
  • Improved version of G.721 and G.723
G.729 8 kbit/s
  • Excellent bandwidth utilization
  • Acceptable quality
  • Encodes audio in 10 milliseconds-long frames with 80 audio samples
  • High compression rate
  • Supports multiple calls simultaneously
  • Royalty-free
GSM 13 kbit/s
  • Global System for Mobile Communications (GSM)
  • High compression ratio
  • Free
  • Same encoding used in GSM cellphones
  • MOS of 3.7
iLBC 15 kbit/s
  • Internet Low Bit Rate Codec (iLBC)
  • Free
  • Used by many VoIP apps, including open source
  • Tackles packet loss, delay, and jitter
Speex 2.15 kbit/s
44 kbit/s
  • Free software speech codec
  • Most preferred for many VoIP apps and podcasts
  • Uses variable bit rate to reduce bandwidth usage
SILK 6 to 40 kbit/s
  • Developed by Skype
  • Available as open-source freeware
  • Base for the newest codec: Opus


What is the Best Codec for VoIP?

While there are a few different voice codecs available, you need to find the VoIP codec that works best for you.

So, which codec is better, G.711 or G.729?

This depends on your business’ bandwidth usage and capabilities, as well as call volumes. But the consensus is that the G.711 seems to offer the most reliable call quality. This codec provides uncompressed high voice quality, but also has high bandwidth usage.

G.729, on the other hand, is the low bandwidth alternative to G.711. However, it may only offer acceptable call quality.

So, it comes down to your specific business circumstances and resources. For these reasons, most VoIP providers accept G.711 and G.729.

Alternatively, G.722 also offers high voice quality, but not all VoIP providers accept this voice codec. So, make sure to ask.

Related: How Much Bandwidth is Needed for VoIP?

Choosing the Right Codec for Voice Calls

Since VoIP codecs need to compress and decompress audio traveling through your phone system, you and your provider must agree on the right codec. In other words, the codec you choose needs to support your team’s bandwidth usage as well as work effectively with your provider’s network.

This means you’ll need to speak with your desired provider to understand their requirements and evaluate that alongside your business.

United World Telecom supports the G.711 codec (both μ-law and A-law), and our call quality has an average MOS of 4.3.

Want to see if we’re a good fit for your business? Call us today at 1 (877) 898 8646 or chat with us online!

What is LAN? Understanding Local Area Networks

Here’s a brief overview of how local area networks work and why businesses need this type of network. Learn about the different types of LAN and what options are available for those interested in VoIP.

What is a Local Area Network (LAN)?

A Local Area Network (LAN) is a group of devices sharing a common communications line to a server within a specific geographical area.

This type of network can serve 2-3 users in a home setting or hundreds of users in an office setting. You can set it up so that end nodes — such as computers and printers — can communicate and share information and resources within your IT infrastructure.

A diagram of a LAN network.

To set up LAN networking, you’ll need:

  • Ethernet cables
  • Layer 2 switches
  • Layer 3 switches and routers (for larger network setups)
  • Desired devices

What Does a LAN Do?

At its core, a local area network allows multiple devices to share a single internet connection.

And it helps users connect to different internal servers and websites. You can also connect to other LANs that reside within the same WAN. By doing this, your teams have access to all centralized applications, which is especially useful for storing business-critical data.

And since this network allows you to connect to different servers and devices (printers, computers, etc.), you can use it for sharing resources.

To enable these connections, you will need both Ethernet and WiFi.


A diagram comparing LAN vs WAN vs VLAN.

Local Area Network Architecture

Now, from an architecture point of view, you can have a peer-to-peer LAN or a client-server LAN.

Peer-to-Peer LAN — connects two devices using an Ethernet cable. These devices could be workstations, PCs, PC-to-printer, etc. This type of LAN does not have a central server, so each device is equally responsible for the functioning of the network. Devices share information through wired or wireless connections.

Client-Server LAN — connects multiple endpoints and servers to a LAN switch, which directs communication between these devices. The server manages storage, application and device access, and traffic while the client connects to the server through wired or wireless connections.

Types of LAN

That aside, there are 3 primary LAN types. Let’s look at them in detail:

Wired LANs

As the name suggests, a wired local area network uses “wires” such as Ethernet cables and switches to connect different devices, endpoints, and servers to the network. Wired LAN is the most common type used in most offices, since it is often very reliable and performs with speed. However, it is not the most flexible or portable network, making it hard to access from different locations and devices.

Wireless LANs (WLANs)

Wireless local area networks connect two or more devices using wireless connections (such as radio transmissions) within a limited area like a home, school, or office. WLAN uses the IEEE 802.11 specification to share data between different endpoints. WLANs offer more flexibility and cost savings since they don’t require extensive cabling and enable connectivity to different mobile devices like smartphones, tablets, etc.

Virtual LANs (VLANs)

VLANs break up a single switched network into a set of overlaid virtual networks. These networks have different functions and security networks. Large LANs at the enterprise level require multiple hardware and software to work optimally. If all these devices are connected to one local network, applications will experience network congestion and slow down.

This is where virtual LANs come in handy. It breaks up traffic so that it is only received by devices within a specific VLAN and not the entire network.

Using VoIP with LAN

You can use a LAN to connect your VoIP devices to your telecom provider using Ethernet infrastructure. This way, when you place a VoIP call, audio signals convert to data packets and travel from your internal network to your carrier’s VoIP network. It is then delivered to the destination and converted back to voice.

So, you will need a reliable local area network and a high-quality VoIP provider for VoIP calls to run efficiently.

The route you choose depends on your resources and network capabilities. Our tech experts at United World Telecom can help you identify the best VoIP solution for your business. Call us today or chat with us online to learn more!