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What Equipment is Needed for VoIP?

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To get the most out of your VoIP phone system, you need the right VoIP equipment to support your business communication needs.

While most newcomers consider VoIP to be expensive and complicated, the truth is that the technology is fairly simple and inexpensive. There is not a lot of hardware or expensive technology required.

In fact, with VoIP, you can convert existing technology into business powerhouses. How?

Let’s have a look at the equipment needed for VoIP.

Must-Have VoIP Equipment in 2024

What equipment is needed for your VoIP system needs depends on the type of VoIP user you are—residential user or business user.

A residential user will need only a PC / laptop, softphone, and a headset. A small business, on the other hand, might need more. Business users will need a web phone or softphone, a computer, a headset, high-speed internet or data, and a business VoIP subscription.

Without the right VoIP equipment, you will struggle with poor VoIP call quality, leading to frustration and miscommunication. To combat this proactively, make sure you get the right equipment and VoIP setup.

Here is a checklist of equipment needed for VoIP to use your service successfully:

1. VoIP Phones: Hard Phones or Softphones

The main VoIP equipment you need is a phone to make and receive calls. Many VoIP and virtual phone service providers offer global call forwarding to route calls internationally to specific locations and numbers. This way, you can divert calls to your VoIP phones or locations as needed.

A VoIP phone uses IP technology (the internet) to transmit calls between two or more parties. There are two types of VoIP phones:

    • Hard phones — are like traditional phones but come with specialized digital hardware.
    • Softphones — are web-based applications or software that can be downloaded on different devices.

You can even use a combination of both types of VoIP phones to support in-office and remote users. Since softphones can be used from any location and device, you can

2. Headsets

Next, you will need VoIP headsets for your users. VoIP hard phones come with a microphone and receivers. However, if your users use a softphone, then they will need a headset with a microphone. While the tendency is to purchase inexpensive VoIP headsets, these headsets can lead to VoIP issues such as latency and network jitter.

Take time to find good quality headsets that are hands-free, remain connected, have noise-canceling technology, and, overall, improve call quality.

equipment for voip

3. Computers or Laptops

Next, you will need personal computers to complement your service. You can use computers and laptops to not only get work done but also make and receive your VoIP calls through a softphone. This way, users can manage calls and update their CRM or task managers as they work without interruption. And, users working remotely can also use their laptops to connect through cloud-based and virtual software and continue working.

4. High-Speed and Reliable Data

Next, if you’re using VoIP for business, then you need a robust internet setup. This includes a reliable and stable internet connection with a modem and router. VoIP does not demand much but needs enough bandwidth to handle your call volume.

While setting up your internet, work with your internet service provider and IT team to set up VoIP QoS for your VoIP service. VoIP Quality of Service (QoS) can help troubleshoot voice over IP issues. When QoS is set up, you can prioritize network traffic to make business calls a high priority. This way, even with everyone using the internet for multiple reasons, the quality of your business calls remains untouched.

learn to set up voip

5. VoIP Phone Service

Finally, you will need a VoIP phone service, and you can get this from any virtual phone service provider. However, you want a provider who offers quality service for a reasonable price and has easy-to-access, responsive customer service. When settling on a VoIP provider, ask about the installation and set-up processes and the different features that come included with the service. Look for customer reviews and case studies to get an idea of how the product works in action. Then, choose a provider that meets your needs and budget.

Get Your VoIP Phone Service with United World Telecom

Now that you know what equipment is needed for VoIP, you can set up VoIP phone service for your business. United World Telecom can get you set up with high-quality VoIP phone service for your business. We have been offering cloud communication solutions to businesses around the world for the past 28 years. Learn more about our VoIP service by speaking with our experts. Call us today at 1 (877) 898 8646 or chat with us online!

What is a SIP Proxy and How Does a SIP Server Work?

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We write a lot about SIP. That’s because we have been a leading SIP trunk provider for over twenty years. Read on to learn more about SIP servers and how they work.

What is a SIP Server?

A SIP server or SIP proxy processes session initiation protocol (SIP) requests. This server is the main element of an IP private branch exchange. SIP is an internet protocol used to initiate and receive voice and video communication by transmitting data packets across an internet connection. This enables the quick and easy transmission of SIP calling between 2 or more parties.

How Does a SIP Server Work?

A SIP server works alongside a voice over IP or VoIP phone system. Both systems together make cloud communications possible. A SIP proxy can:

  • Set up a session between 2 or more endpoints; such as audio or video conferencing between 2 or more parties
  • Replace one endpoint for another; during call transfer or routing
  • Negotiate and adjust media parameters and specs during a session; such as putting a call on hold
  • Terminating a session

It is important to note that the SIP server does not actually transmit media. Media transmission is performed by a media server using the RTP protocol. Within an IP-PBX, the SIP server and media server are present on the same machine. However, a high-volume SIP server like a VoIP provider may separate the two servers on different machines and balance the load.

Additionally, there is no fee or charge to get a SIP address for your server. These addresses connect to unique phone numbers. This enables each user on a SIP network to have a direct inward dialing number to place calls. Furthermore, companies can use these systems in a package such as a hosted PBX.

SIP Proxies: Modes of Operation

A SIP server generally operates in one of two modes: Stateless or Stateful.

1. Stateless SIP Proxy: This type of SIP proxy receives and transmits messages but does not keep any record of the transmission. A stateless SIP proxy works this way: Send > Receive > Delete. This server works at a faster speed because of its limited functionality. Additionally, this simplicity in functioning makes it desirable to small businesses as they can easily scale and upgrade their SIP system.

2. Stateful SIP Proxy: This type of SIP server transmits as well as stores messages and information to access later. Because of this functionality, it can pick up a request message and try again. Or, it can reroute the message through another aspect of the network. A stateful SIP proxy works this way: Send > Receive > Save. An example of this is Time of Day Routing that routes incoming calls based on the time of day and predetermined rules. For example, calls made to a business after hours can be forwarded to a different office location or remote agent.

What Does SIP Trunking Do?

SIP trunking is a service that enables your PBX system to send VoIP and SIP calls over the internet. This service works with virtual telephone lines and sends and receives messages through bandwidth data. You can get multiple SIP trunks and cover various geographic areas. SIP trunking makes it possible for your business to expand operations beyond your immediate location.

An image of a SIP proxy and server.

Benefits of a SIP Proxy and SIP Trunking

SIP servers and SIP trunking have become increasingly popular with businesses of every size. Here are the top benefits of switching to SIP:

1) Enable Unified Communications
SIP trunking enables voice, video, and text messaging from one platform. There’s no need to invest in different services to keep your communications stable. You can make and receive high-quality calls, audio and video conferencing, and texting from SIP trunking.

2) Forward Calls
With a SIP server, you can quickly and easily forward or direct incoming calls to several SIP and VoIP devices. This is helpful for any office with a busy call volume. Plus, with a stateful SIP proxy, you can save and access calls or messages that didn’t go through the first time and try again later.

3) Cut Communication Costs
VoIP and SIP are in demand because it not only offers high-quality communication but also a comfortable price. Sending and receiving voice, video, and text over a SIP server costs almost nothing to users.

4) Network Security
Secure VoIP is a necessity within any business. A SIP server protects your communication system from hackers by disconnecting calls and users without credit or authorization.

5) Access to VoIP Features
With SIP trunking, you can gain access to useful voice over IP features that can help organize calls and provide a professional image to your business. Features include, but are not limited to;

  • Call forwarding and routing options
  • Automated greetings
  • Analytics and reports
  • Unlimited extension, and so many more

Where Can I Get SIP Trunking?

United World Telecom can help! You can buy SIP trunks directly from our website or contact one of our experts to learn more.

What is VoIP QoS and How to Set it Up?

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Are you noticing more dropped calls or jumbled audio during your business VoIP calls? While cloud phone services can be much superior to traditional phone services, they cannot guarantee high-quality phone service at all times. This is because many factors influence the quality of a VoIP call. Here we discuss the importance of VoIP QoS and how to set it up.

VoIP QoS: Definition, How it Works, Best Practices

VoIP Quality of Service (QoS) is the process of addressing and fixing voice quality issues so your employees can effectively communicate with business contacts. This process is crucial to understanding how your business phone system works and what you need to do to improve it. QoS is one such way to maintain good call quality.

What is VoIP Quality of Service (QoS)?

VoIP QoS prioritizes network traffic passing through a router to provide acceptable-to-good service to users. Quality of service, therefore, helps address voice quality issues within a virtual phone system.

In a VoIP phone system, voice protocols are converted into data packets and transferred between two or more users. However, in order to obtain high call quality, these data packets need to arrive at the destination together and in the right order. And that’s where VoIP QoS comes into play as it sorts out traffic by dedicating resources where needed.

Is QoS needed for VoIP?

VoIP calls can be prone to jitter, latency, and packet loss issues which lead to bad audio quality, network congestion, missed or jumbled audio, dropped calls, etc. And for any business, this is a major issue because it interferes with your teams’ ability to offer uninterrupted service. Important business communications revolve around:

  • Sales calls and demos
  • Client and lead generation
  • Customer success and customer support
  • Technical and IT support
  • Remote team management
  • Internal business meetings
  • Meetings with clients and business contacts
  • Employee interviews

Network congestion can easily affect the quality of these calls, which in turn, leads to misunderstandings, frustration, low brand trust, low productivity, and more. Without QoS, VoIP calls are not functioning optimally. And that must be addressed.

Troubleshooting VoIP Call Quality with QoS

So, what does QoS do for VoIP phone systems? Routers generally handle data packets via a ‘first in, first out’ order. And all traffic is given the same level of priority. This means that all traffic from every device (phones, computers, mobile phones, etc) on a network get the same level of priority.

When bandwidth usage is high — too many people using the internet — these data packets can end up in long queues. Call quality competes with other applications running on the network. This is true for employees working within an office or remotely.

So, naturally, this affects voice call quality and successful VoIP deployment.

VoIP Quality of Service helps balance out how much bandwidth is needed for a certain type of traffic or certain devices. In other words, it spreads out the available bandwidth across devices and applications to ensure the right traffic gets through hassle-free.

VoIP QoS Settings & Requirements

So, what does your business need to do to ensure high call quality? Here, we will go through QoS settings and requirements to be aware of and best practices your business can implement.

VoIP Standards for High Call Quality: QoS Requirement

In order to improve VoIP call quality, you will need to set up QoS accurately. For good VoIP call quality, Cisco suggests these VoIP standards:

VoIP Quality of Service Standards

Understanding QoS Settings

Your business needs to troubleshoot VoIP issues to ensure optimal service. Let’s take a closer look at what these recommendations and settings mean.

DSCP EF
DSCP is short for Differentiated Services Code Point and EF is short for Expedited Forwarding. This model is designed to provide resources to reduce latency (delay) in traffic. In other words, this value sorts through traffic to prioritize VoIP traffic, leading to better, less interrupted call quality.

Packet Loss
Packet loss is the number of packets that get lost during transmission. More than 3% packet loss means low audio quality. Faulty routers, loose cables, and wires, low bandwidth, or poor WiFi signals can lead to packet loss.

Latency
Latency is the delay between the time the speaker speaks and the receiver hears them. Fix latency to avoid users speaking over each other or missing important pieces of audio.

Jitter
Jitter refers to a data packet arriving after the rest of the packet. Data packets may take different paths to reach the destination. This way one or two packets may reach later than the others and this can cause missed calls or jumbled audio. Fix jitter to ensure all packets arrive accurately and voice quality is maintained.

Bandwidth
Bandwidth for VoIP refers to the rate at which data is transmitted or transferred through an internet connection. Lower bandwidth leads to slow speed and low VoIP call quality.

How to Set Up QoS for VoIP?

The first step to setting up VoIP QoS for your business phone system is to reach out to your internet provider and VoIP service provider. These sources can provide exact guides on how to prioritize voice traffic.

Next, implement those suggestions and other call quality best practices. Work with your IT team to make sure everything is set in place for optimal call quality. You may even educate your employees on using other services (streaming services) while working.

VoIP Call Quality Best Practices

Here are the top VoIP call quality best practices to ensure your phone system works efficiently and calls are not interrupted:

1. Check and upgrade your internet speed: Make sure your network has enough bandwidth to handle call traffic. Regularly test your internet speed.

2. Upgrade cable connections: Use wired Ethernet connections such as a Category 6 certified cable. These cables can offer low latency and can support high levels of phone communication and data transfer.

3. Check network and equipment configuration: Network and equipment that are not appropriately configured can lead to disruptions and low call quality.

4. Test network connections regularly: Check ping, jitter, latency, and bandwidth regularly for network congestion.

5. Upgrade your VoIP provider: If your service is still affected despite troubleshooting, then it may be time to find a new phone service provider. Research providers to understand what services they offer and how they can help with VoIP call quality issues.

Get VoIP for Business With United World Telecom

United World Telecom offers VoIP phone numbers and services to businesses around the world. Whether you are a small business looking to expand or are a remote company, we can help you create a robust phone system with high voice quality and reliability. Call us today to learn more about our solutions!

9 Ways to Minimize VoIP System Downtime

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One of the top benefits of using VoIP for business communication is that these systems are reliable. With proper redundancy, you can ensure that if your phone system incurs an issue in one location, calls can be routed to another location or line. This way, you can ensure business continuity and minimize VoIP system downtime.

Here we will look at the different ways you can minimize phone system downtime to ensure your business continues operating even during an internet outage or disaster.

How to Minimize VoIP Downtime During an Internet Outage?

VoIP phone systems have consistently proven to be more reliable than POTS lines. However, since VoIP systems work over the internet, they still run the risk of downtime if somehow your system loses connection to the internet.

Keeping Your VoIP Phone System Running Efficiently

With VoIP systems, you can set up backups so your business communication does not suffer during an outage. Service outages do not have to disrupt your business phone calls. You can either wait it out and hope it restores on its own, or you can prepare in advance to minimize downtime. Here are some ways to minimize and prevent phone system downtime:

To Prepare and Prevent Before an Outage

  • Choose the Right VoIP Provider
  • Set Up Automatic Call Forwarding and Routing
  • Invest in VoIP Monitoring Services
  • Route Incoming Calls to Other Locations
  • Have a Backup ISP

To Do After an Outage

  • Check and Confirm the Power Outage
  • Connect to a Backup Battery Power Supply
  • Use VoIP on Your Smartphone
  • Divert Calls to Voicemail or Other Locations

How to Prevent and Minimize Phone System Downtime

So, what can you do to minimize VoIP system downtime and keep your VoIP phone system running during an outage or disaster?

1. Choose the Right VoIP Provider

It is crucial that you choose a VoIP phone provider that meets your needs but also promises reliability. Research a provider’s record for reliability and the options they offer in case of an outage. Ask your provider what steps they have in place to prevent and minimize phone system downtime, especially during an outage or disaster. Ask about their security policies. You also want to check where your provider hosts your VoIP systems in safe locations.

2. Set Up Automatic Call Forwarding and Failover Routing

A popular way to minimize VoIP system downtime is to forward incoming calls to different locations or devices when your main VoIP phone system is unavailable. This way, during an outage or disaster, incoming calls automatically route to offices and employees in other locations or mobile devices.

With United World Telecom’s VoIP service, you can set up call forwarding with various failover rules for each number or line you have.

3. Invest in VoIP Monitoring Services

It is also a good idea to use a third-party VoIP monitoring service. This service keeps tabs on the status of your VoIP phone system. It will look for and identify potential network problems that may cause disruption. By monitoring your system and alerting users of issues, you can quickly troubleshoot any issues and get back to work.

How to minimize VoIP downtime.

4. Route Incoming Calls to Other Locations

When you use simultaneous ringing or sequential call forwarding features, you can forward calls to different locations when needed. With simultaneous ringing, you can ring multiple phones with one number. And with sequential forwarding, you can ring calls down a predetermined list of numbers — when the first line is busy, the call automatically moves to the next person in line. This way, when your main office or service center is not available, calls are routed and rung on other lines in different locations. This is another good technique to reduce VoIP system downtime.

5. Have a Backup ISP

To protect from internet outages, it is also recommended to have a backup WAN provider. Since VoIP services only need a low-latency connection, any broadband connection should work. You can find a high-speed wireless provider as a backup, or use 4G LTE technology from a cellular provider.

How to Minimize VoIP System Downtime After an Outage

While there are ways to prepare for an outage, and even prevent one, certain situations may be out of your control. So, what can you do to recover from an outage and get your communications up and running?

6. Check and Confirm the Power Outage

First, check to make sure you have a power or internet outage in all rooms and among all devices. Check the electric service panel, if you have a circuit trip, flip the circuit breaker to ON and everything should be working again. You may even check what the electric and internet situation is with neighbors and look for updates on local outages.

7. Connect to a Backup Battery Power Supply

Connecting to a backup battery source is a temporary solution but gives you enough time to identify the issue and resolve it. You can use an uninterruptible power supply (USP) and connect your WiFi system and phone system to it to restore connectivity and power. Note that this solution only works if your ISP is not affected by the power outage.

8. Divert Calls to Voicemail or Other Locations

Using your smartphone, access your VoIP provider’s control panel, and adjust call forwarding settings. You might choose to route calls to your voicemail box or to employees in other locations. You can add in multiple numbers and SIP devices that these calls will forward to. This way, callers trying to reach your business can still get the assistance they need. You may even choose to add a new greeting or pre-recorded message letting customers know that your phone system is down but they can leave a voicemail message, and your reps will get in touch with them as soon as possible.

9. Use VoIP on Your Smartphone

Lastly, if your power or internet outage does not affect the use of your smartphone, you can use your VoIP phone service through the smartphone as a temporary solution. For example, you can download our softphone app on your smartphone and make and receive business calls with your business phone number(s). With our softphone, you can:

  • Update customer and contact information
  • Transfer calls to other users
  • Make outbound calls with dynamic caller IDs
  • Access voicemails, and more

Do You Have a Backup Plan in Place?

The best way to minimize VoIP system downtime is to prepare for it. The above are some ways you prevent loss of data and usage by setting up backups in place. Want to learn more about how our VoIP phone service has 99.999% uptime? Call us today and chat with our representatives!

Top Business VoIP Trends to Know in 2025

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Technology continues to advance over time with new and improved options available to users of all kinds. And for businesses, this is always exciting because there are new ways to improve customer experience (CX) and scale vertically. One of these technologies that impact businesses the most is business VoIP.

Future of Cloud Communications: 2025 VoIP Trends

VoIP, a type of cloud communications, makes it easy for users to communicate through the internet securely and reliably. And VoIP providers are developing new ways to increase their offerings and provide businesses with a well-rounded phone system.

So, what’s new in VoIP? Quite a lot: from leveraging AI to UCaaS and CCaaS to 5G networks. If your business is trying to catch up, you may want to consider an upgrade to your phone system to help improve CX and customer satisfaction.

Here are the top 9 business VoIP trends in 2025.

1. Unified Communications as a Service

One of the top business VoIP trends in 2025 includes bringing all communication and collaboration needs into one platform. This becomes crucial for businesses with multiple offices and locations around the world and for those with remote workers. Unified communications (UC) and cloud communication tools improve a business’ ability to be flexible and mobile and scale as needed. With collaborative features and real-time updates, your teams can stay organized and productive, no matter the location.

2. Mobile Unified Communications

Besides working within your business phone system, UC services are also being extended to mobile. The rise of mobile UC means you can transform your mobile or smartphone into a business powerhouse. You can use this device then to conduct both personal and business interactions and give your business more mobility. Mobile VoIP systems give you access to smartphone and PC apps, which increase productivity on the go.

Related: UC Trends You Need to Know in 2025

3. Advanced Video Conferencing Services

Since the COVID-19 pandemic led to many businesses switching to work-from-home or remote work capabilities, video conferencing became crucial for the proper functioning of many businesses. Video conferencing tools were in high demand and that competition was troublesome for VoIP service providers. This led to one of the newest VoIP trends: video conferencing. VoIP providers are upping their game and now offering multi-channel communication services, including video conferencing. They do this through integrations and partnerships with video conferencing services. And while this is the current solution, evolving technology will open doors for advanced VoIP software with more conferencing and collaborative capabilities to make doing business over the cloud easier and more user-friendly.

4. Customizable Features

Each business is unique and follows specific processes and guidelines that work best for the business and its employees. The same goes for your business phone system. Cloud communication is known greatly for its customization capabilities. Need to upgrade your system, add new lines without interrupting service, or integrate with other services? Cloud communications and VoIP systems can help you create a business phone system that works best for your business needs. Customize your system for the number and type of users you have, where they are located, different departments, and so on.

VoIP trends to watch in 2021.

5. Seamless Integrations

VoIP integrations can help you expand your VoIP phone system’s capabilities. You can connect your office calendar, CRM, content management systems, lead generation platforms, video conferencing services, chatbot automation, and more. Most popular integrations that work well with cloud communication platforms include:

  • Salesforce
  • HubSpot
  • Azure
  • OneLogin
  • Zapier
  • Clio

6. Leveraging AI

As seen in recent years, the use of AI within business has increased. This goes for the business VoIP world too. From handling customers to automating responses and ticketing systems, AI has been used to improve customer experience and satisfaction. Most companies are using AI to automate their websites and communication platforms to streamline services and improve engagement. This VoIP trend will continue in the coming years as AI has exceptional potential. Common uses of AI for VoIP include:

7. Security Issues

For the longest time, users considering VoIP were concerned about internet security and cyberattacks. But VoIP and internet services have come a long way and can provide secure avenues for their services. Business communication involves customer and company data and privacy which needs to be kept safe from intruders and hackers. VoIP security ensures that your business can communicate safely with its customers and employees. Learn more about how VoIP tackles security concerns in our guide to VoIP security.

8. 5G

In our fast-paced world, we expect quick response times and processes. And businesses need to meet these expectations when dealing with customers. 5G networks — one of the most anticipated VoIP trends — are here to solve this very issue and to improve VoIP call quality. With 5G, your business can:

5G networks are built to improve customer satisfaction and CX by making your business more available and responsive.

9. VoIP in More Places

More and more businesses are switching to VoIP to improve their communication and customer outreach. From schools to small businesses to large enterprises, every type of business can benefit from VoIP services. This makes business VoIP more present in more places around the world.

Need a VoIP Upgrade?

Want to learn about United World Telecom’s VoIP number solutions and how we can help your business improve internal and external communication? Call us at 1 (877) 898 8646 or chat with our experts today!

Cloud PBX vs On-Premises PBX: Which is Right for Your Business?

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When evaluating your options for a private branch exchange, you will often come across the terms: cloud PBX and on-premise PBX. What is the difference between the two services and how do you decide which one you need?

Getting the Right PBX System for Your Business

A PBX, or private branch exchange, is a private telephone network. A PBX system makes internal call management easy and ensures smooth collaboration between different teams and departments within a business.

There are two common types of private branch exchange systems: Cloud-Hosted and On-Premise. And choosing between these two systems depends on a few factors:

  • What do you need from your phone system?
  • Do you want to connect multiple office locations through the system?
  • What communication features do you need?
  • What is your budget?
  • Do you have or need to hire an IT team?
  • The answers to these questions can help you determine which phone system is right for you.

What is a Cloud Hosted PBX?

A cloud PBX is what it sounds like: a private phone exchange that works over the cloud. This means that it does not need a physical space in your office. The host will take care of the maintenance of the system and all you need to do is use it.

You will not have to worry about installing any hardware, saving office space for the system, or high maintenance costs. You won’t even need an on-premise IT team to handle all of it. However, you might not have as much control as with an on-premise system.

How Does an On-Premises PBX Work?

An on-premise PBX system is installed on company premises. This type of phone system is installed, run, and maintained by your company. You might need an experienced IT team to handle such a system.

Choosing Between Cloud PBX vs On-Premises PBX

The bottom line is that if you want a system that is wholly controlled and managed by your in-house IT team, then an on-premises PBX is the choice for you. However, if you don’t want to worry about the hassle of running and maintaining your phone system, then go with the system that’s hosted in the cloud.

Here are the other main differences between the two:

 

Cloud Hosted PBX On-Premises
Hosted by your provider Operated on-site by you
Little to no control over how it operates Controlled and managed fully by you
No physical space needed Physical space needed
No installation or maintenance needed Needs to be installed and maintained regularly
No professional IT experience needed Needs an experienced IT team to operate the system
Uses VoIP tech; Needs a broadband connection May use VoIP tech
Low monthly costs; No installation costs Low monthly costs; High installation costs
Variety of virtual calling features Limited calling features
Scalable as needed Limited scalability
Recommended for small to medium-sized businesses Recommended for large corporations

United World Telecom Knows About PBX Systems

United World Telecom has the PBX solutions you need for your business. You can also choose to set up your own IP-PBX system. This way you can take full advantage of SIP to place calls over the internet. Doing so results in better international calling rates at higher scale.

What is SIP Calling?

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Communicating effectively with customers is crucial to the successful running of any business. It is through your customers that you can improve your product and increase sales. And so, if your business isn’t doing everything it can to make it easy for customers to connect with you, then you are falling behind. Learn what SIP calling is, how it works, and how businesses can benefit from using a SIP phone system for business communication.

SIP Calling: The What, How, & Why

To understand SIP calling (or SIP trunking), you must first be familiar with Session Initiation Protocol (SIP). SIP is a signaling protocol that initiates real-time voice calling between two or more parties. This IP is in charge of starting, maintaining, and terminating the call over an internet network. By doing so, SIP technology makes it possible for users to make and receive high-quality calls over a virtual network. Let’s look at how SIP calling works and how businesses can use SIP for improved communication.

What are SIP Calls?

SIP calls use Session Initiation Protocol to transmit voice calls over a SIP trunk or SIP channels. In other words, SIP calls are voice calls sent over an internet protocol or internet connection.

Often used interchangeably with voice over IP or VoIP calls, the two systems are different. VoIP makes SIP calls possible. This is because SIP uses VoIP technology to transfer calls from one end to another destination over a stable internet connection.

How Does a SIP Call Work?

Traditional phone systems consist of a PBX system and phone lines connecting to a PSTN.

SIP technology removes the need for a traditional, physical connection. With SIP, you do not need to be connected to a phone company or geographical location. And, you do not need multiple phone lines for different departments. You will get a SIP trunk, instead, to run your virtual phone system. You can then establish voice communications virtually via the internet.

And what’s the end result? You can get a virtual phone system with call management and call routing features (global call forwarding, outbound calling, etc.) without physical or multiple phone lines. This system can be used from any location, connecting multiple devices.

sip calling for business

Why Does Your Business Need SIP Calling? 5 Benefits of SIP Calls

So, how can SIP calling benefit your business? From being a cost-effective alternative to creating a unified communications platform, SIP trunking can help businesses organize their internal and external communications to connect better and increase productivity and efficiency. Let’s look at the top benefits of SIP calling for business:

1. Save on Business Calling Costs
First and foremost, the cost of SIP calling is highly affordable for businesses needing multiple phone lines and with various departments. Not only can you make and receive high-quality voice calls inexpensively, but you can also bypass international calling rates when offering global customer support.

2. Scale Up or Down as Needed
SIP trunks are designed to support a business’ scalability needs. This means that if you need to scale up and add more direct inward or outward dial numbers, you can do so easily. And the same goes for scaling down; that is, removing lines or pausing certain services and features. And most of these actions can be done by you, reducing the number of times you will panic-call your SIP provider.

3. Improve Communications with Better User Experience
SIP trunks, and SIP technology generally, are easy to use. You do not need to employ new IT teams or conduct rigorous training or worry about setting up complicated software. You and your employees can simply manage everything from a user interface provided by your SIP provider. Use this interface or control panel to set up features and service, make changes or adjustments, add lines, and more.

4. Experience High-Quality Voice Calls
When using SIP, your business no longer relies on physical landlines. This means that your communication system does not easily fall apart due to power outages or weather conditions. SIP calling utilizes redundancy to automatically reroute calls from one location to another if the previous location’s user is unavailable or inactive. This is a reliable way to make and receive high-quality business calls.

5. Offer Excellent Customer Service
Never miss calls by routing them to different locations, in case the first is unavailable. Reduce the number of dropped calls or low-quality calls. All of these factors come together to help your business offer excellent and uninterrupted customer service to not only local but global customers as well.

Get SIP Calling with United World Telecom

With a SIP phone system in place, your business is gearing up to communicate with your customers conveniently and cost-effectively. Learn more about how SIP trunking can boost business communication by chatting with our experts or calling us at 1 (877) 898 8646.

What Is SIP ALG and Why VoIP Users Should Disable It

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In this post, we discuss what SIP ALG is and how it can affect the quality of your VoIP phone calls. Disable SIP ALG to improve VoIP call quality and ensure there are fewer interruptions.

What is SIP ALG?

Session Initiation Protocol (SIP) is an internet protocol with voice data packets that initiates, maintains, and terminates voice communication between two users. SIP is used for voice calling over LTE and VoIP phone systems.

Routers used to connect to the internet also segment the provider and your internal network through Network Address Translation (NAT). This is to add an additional layer of security through a firewall allowing only authorized systems access as they connect with a network’s computers and devices.

The main purpose of SIP ALG — Application Layer Gateway — is to prevent problems caused by a router’s firewall. ALG prevents these issues by keeping an eye on the VoIP traffic (voice data packets mentioned earlier) and modifying them, when necessary. ALG works as a proxy to rewrite the destination for these packets. By doing this, ALG can improve connectivity.

Why VoIP Users Should Disable SIP ALG

Many routers have the SIP ALG feature turned on by default. With this feature on, VoIP traffic (voice data packets) can get lost due to router firewalls when transferred between the phone and the VoIP provider.

And because of this, it can lead to multiple VoIP problems, including:

  • One-way audio
  • Phones not ringing on incoming calls
  • Calls sent directly to voicemail, especially when not set to do so
  • Dropped calls, even after connecting

This is why one of the best ways to improve VoIP call quality, among others, is to disable the SIP ALG feature.

SIP ALG phones
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How to Disable SIP ALG in your VoIP System

Disabling SIP ALG is quick and easy, and depends on the type of modem your business uses. For most routers, you will need to:

  • Log into your router’s control panel.
  • Navigate to Advanced or Security settings.
  • Locate SIP, ALG, or Firewall settings (depends on your router’s set-up).
  • Uncheck the SIP or ALG box.
  • Save and reboot/restart your router.

If your router’s settings are not as clear, you can always reach out to your provider and ask for specific instructions.

Protect and Maintain VoIP Call Quality

Disabling SIP ALG is a common way of troubleshooting VoIP issues. However, there are other VoIP call quality issues such as jitter, packet loss, and latency that can affect the way your business communicates with its customers. Most of these issues stem from low-quality internet or insufficient bandwidth. Speak with our representatives today to learn how your internet bandwidth can affect your VoIP phone system. Call us at 1 (877) 898 8646 or chat with us online today.

Troubleshooting the 7 Most Common VoIP Issues

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Having VoIP problems and don’t know how to solve them? Here we go over troubleshooting for the 7 most challenging VoIP issues.

7 Common VoIP Troubleshooting Problems

VoIP phone systems help businesses save about 50%-75% of communication-related costs. This is because such systems offer flexibility, mobility, and scalability which helps users connect from any location and communicate through advanced technology.

However, even VoIP phone systems — with their advanced features, high voice quality, and more — are not devoid of possible quality issues. Thankfully, most VoIP call quality can be improved without IT help so you can continue communicating effectively.

The Ultimate Guide to VoIP Troubleshooting [2022]

Is your VoIP system not working? Here are the most common VoIP issues and a simple guide to troubleshooting them.

1. Inability to Make Calls from a Device

Struggling to make VoIP calls from your device? An inability to make calls can be due to a failure to connect, inadequate internet support, and more. For some businesses — like a call center — not being able to make outbound calls to customers and leads can essentially shut the business down until you find a solution.

Most likely, the cause of this problem is the SIP ALG feature is turned on, on your router. Session Initiation Protocol Application Layer Gateway (SIP ALG) is a common feature in commercial routers and is turned on by default. The main task of a SIP ALG is to reduce or prevent issues resulting from router firewalls. It does so by constantly inspecting your VoIP call traffic. However, SIP ALG may modify packets (voice signals) in unexpected ways, leading to problems such as incoming and outbound calls failing and phones not registering.

Solution: A simple solution for outbound VoIP calls failing would be to turn off the SIP ALG feature. If you still experience the issue, then try repositioning the VoIP devices onto a VLAN.

2. Dropped Calls

One of the most common VoIP problems is dropped calls. This causes a lot of frustration, especially during business calls. This is when the call suddenly ends mid-conversation without the speakers hanging up. Call centers or large enterprises with large call volumes face this issue the most.

Solution: First, ensure all devices, software, and hardware associated with your VoIP phone system are updated and running on the current version. If you are still experiencing the issue, disconnect all devices and turn them back on one at a time.

This may be time-consuming but it will help you identify exactly which device is the root cause of the problem. Speak with your small business VoIP provider if you notice that calls get dropped after a certain amount of time. They may have an automatic disconnect feature to ensure calls are not left open by mistake.

3. Jitter

Jitter is one of the most common VoIP problems. Network jitter directly affects voice quality and communication, leading to jumbled, muffled, or missing audio. As voice data packets travel from one destination to the next, some packets may arrive before the other. This leads to out-of-order or missing parts. If such voice quality occurs for more than 30 milliseconds then the overall call quality is impacted. As such, when finding a new provider, look for one that can keep the delay under 20 milliseconds.

Solution: Your internet may not have enough bandwidth for VoIP. Upgrade your internet connectivity by contacting your ISP.

Fixing VoIP issues at the tap of a button.

4. Echo

This is a pretty straightforward VoIP concern. Telephone echo leads to voices being repeated at various points, leading to confusion and possible miscommunication. Often the recipient of the call hears the echo while the caller may or may not be aware of this VoIP problem. Echo can be a result of either feedback during the conversation or a VoIP phone system issue. As such, it can be troublesome when conducting important business calls such as conferences, sales, and support calls.

Solution: First, if your phone is using the speaker option, take the call off the speakerphone. When using a speakerphone, the voice has to travel through multiple microphones and speakers leading to disruption in the audio for the recipient. Additionally, you may even need to test the phone headset you use and consider getting a high-quality replacement. Lastly, echo can be a result of a bad internet connection or inadequate bandwidth. Check your speed with an Internet speed test and also reevaluate your wall jacks, Ethernet cords, and other cables to ensure there are no damages.

5. Broken/Muffled Audio

Broken, muffled, or choppy audio refers to words and audio being dropped, interrupting calls when connected. This is one of the most common VoIP issues users face. Thankfully, it has a solution.

Solution: How you solve the problem of broken audio depends on who is experiencing it. If your business is experiencing the issue, it is most likely due to insufficient bandwidth that leads to packet loss as all voice packets are transferred successfully. A common VoIP troubleshooting solution for this problem is to turn off other applications that take up a lot of network space and are not needed for business. This includes streaming services like YouTube or Netflix and so on. Additionally, make sure your router’s Quality of Service (QOS) settings have the VoIP service on priority.

6. No Sound

Similar to broken audio, a voice call with no sound after connecting can lead to frustration and interruptions in communication. No sound in a voice call can be a one-way issue (where one party hears but others can’t) or a two-way issue (where both parties cannot hear).

Solution: One reason for a lack of sound during calls may be because of firewalls blocking RTP packets. Examining and possibly disabling your SIP ALG can solve this problem.

7. Phone Doesn’t Ring on Incoming Call

This VoIP issue is pretty straightforward: missing calls from important customers and clients because the phone doesn’t ring. Another version of this issue is if your calls are sent directly to voicemail instead of an employee.

Solution: Thankfully, this common VoIP problem has an easy solution. First, ensure your device is registered within your VoIP phone system and VoIP provider. Also, check to make sure your device is not on the Do Not Disturb setting and has the correct call forwarding settings and configurations.

Choosing a Reliable VoIP Provider

If your VoIP system is not working, you might need a new, high-quality system that comes that comes with easy problem-resolution.

Finding a reliable VoIP provider for your small business can be tough if you do not know what to look for. United World Telecom has been in the business for over 25 years and we offer top-quality communication services.

Try our VoIP phone system solutions! Call us today at 1 (877) 898 8646 or chat with us online!

What is a RespOrg?

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A quick guide to what RespOrgs are, how they work, and how businesses can benefit from using a RespOrg service provider for their toll free numbers.

RespOrg: Definition

A Responsible Organization or RespOrg is a company (usually a telephone company) that is certified to have access to a centralized database of toll free numbers. This centralized database is known as the 800 Service Management System or SMS/800.

How do RespOrgs Work?

RespOrgs are in charge of managing toll free databases, assigning numbers, and keeping records. If you want a toll free number for your business, you will need to contact a RespOrg. For customers, RespOrgs come into play when porting a toll free number. To port a toll free number, a current user will have to change the RespOrg ownership from the old carrier to the new carrier. You will need a Letter of Authorization from your new carrier and your current RespOrg must authorize the release of the number to the new RespOrg or carrier.

A business that has high toll free traffic can take advantage of one of the below choices:

  • Become their own RespOrg
  • Use a single carrier for all of their call volume
  • Use a RespOrg service provider

Port your toll free number to United World Telecom.

How to Become a RespOrg?

RespOrgs can be large or small companies or even run by a solo business owner. Some toll free number carriers or business phone service carriers may also be RespOrgs. Currently, the US has about 400 RespOrgs. To become a RespOrg, a business goes through a certification process.

Technically, any company or organization that uses a toll free number can become a RespOrg. To become a RespOrg, your business will need to do the following:

  1. Complete and submit a ten-page service establishment form
  2. Pay a deposit (avg. $4000)
  3. Pass a certification exam to be certified

Should My Company Become a RespOrg?

While becoming a RespOrg is an easy process, there are a few factors to consider. For example, you will need to factor in the salary of the employee managing the toll free traffic. The cost of being a RespOrg for your business — as opposed to using a RespOrg provider — may entail increased expenses. Plus, if your employee leaves, you will need to train and certify a new employee, which will require additional costs.

Many businesses, therefore, choose to work with a RespOrg service provider to reduce costs. RespOrgs will work with your business and your specific needs to offer you the best pricing. Some benefits of using a RespOrg include:

  • Ability to route calls to different carriers
  • Routing calls at different times of the day
  • Taking advantage of low-cost carriers in different countries
  • Access to Disaster Recovery — in case your toll free carrier is shut down, traffic can be routed to a secondary carrier

A company with high toll free traffic will find it beneficial to utilize a RespOrg service provider instead of becoming their own RespOrg.

Get a Toll Free Number for Your Business Today!

United World Telecom offers toll free business numbers for more than 160 countries around the world. You can get a toll free number to enter new markets and extend sales and customer support services to more customers. We also offer number porting services for businesses that currently have a toll free number but are not satisfied with their service. Sign up on our website today or call us to learn more!

PRI Explained: What is a Primary Rate Interface?

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Choosing a business phone system for your company is a necessary part of creating the perfect communication system. With advancements in technology, there are many different systems available for businesses to choose from. Here we will discuss primary rate interface (PRI) and the advantages and disadvantages of this phone system.

What is PRI?

A primary rate interface or PRI is a communication system that is provider-free. This system allows businesses (users) to send and receive voice, data, and video files through a copper wire network. PRI systems or lines constitute two pairs of copper wires. This feature of primary rate interface networks provides secure data transmission. You can get two types of PRI systems:

  • Basic rate interface solutions (BRI) for personal and small business use
  • PRI for large enterprises and corporations.

Features of Primary Rate Interface

To understand how these communication systems work, it is first crucial to be aware of their features. Key features of a PRI system include:

  1. Lines are made of two pairs of copper wires connecting the provider and the user.
  2. You can have 23 B-channels on a single telephone line. And by doing so, it enables businesses to have multiple extensions and telephone numbers via one connection.
  3. Each channel has 64 kbps for data transmission.
  4. Can connect two private branch exchange or PBX systems together and can also work with an IP PBX system.

Advantages of a PRI Phone System

There are different ways a primary rate interface phone system benefits businesses. However, whether or not your business needs this system depends on what you hope to achieve through your business communication system. Let’s look at how PRI systems boost business communication:

1. Extensions and DID numbers:

Direct inward dialing refers to direct numbers assigned to individuals within a business. This means that callers from outside can dial this number and reach a contact directly. Extensions work in a similar way with an additional code attached to a number to let callers reach an individual or department directly.

With PRI, SIP trunking, or virtual phone systems, you do not need additional lines for each number or extension. For PRI, specifically, you can have up to 23 conversations happening simultaneously on one line. That means you can have up to 23 users using the system. And that is considering everyone uses it at the same time. If you need simultaneous communication, you can add more users to these existing lines and they can use it as and when needed.

2. Scalability and expansion:

As your business traffic grows and communication needs increase, you will want to scale and expand. And a primary rate interface will allow you to do that. If more users are needed, you can simply get another PRI line and add it to your existing system, giving 23 more users the ability to communicate.

PRI vs hosted voip

PRI Drawbacks

While a primary rate interface system changed the way businesses communicated over the years, phones have come a long way since. Advancements in telecom technology have given rise to more modern and user-friendly systems.

The biggest drawback that PRI systems have is the ability to expand in bundles of 23. This means that if you have just one or two extra employees and all channels are used constantly, then you will need to buy 23 more channels for those extra employees. You will end up paying more than you need.

On the other hand, if you run a large corporation with 100-150+ employees, then you will need multiple PRI lines to work efficiently. Additionally, it gets more complicated if you need to add multiple locations or remote workers.

To combat these issues, you have a few alternatives to consider: Hosted VoIP and SIP trunking.

PRI vs Hosted VoIP vs SIP Trunking

Most businesses today have adopted a cloud VoIP or hosted VoIP solution. Hosted VoIP means that your service provider hosts your phone solution and takes care of all your software needs and maintenance. All you do is use the service. You do not have to worry about purchasing hardware and software, maintaining it with a professional IT team, and so on. This helps your business save on communication and IT-related costs.

SIP trunking is a session initiation protocol (SIP) feature that enables transmission of voice communication over a data network. SIP trunking works similarly to POTS except that the phone lines are virtual instead of standard copper lines. And your phone system connects to your provider via your internet connection. SIP trunking has often been used as an alternative to POTS and PRI systems.

PRI, unlike VoIP and SIP trunking, does not rely on internet bandwidth for transmission, and therefore does not suffer from jitter or packet loss. However, there are limitations in terms of scaling upwards, mobility, and features available.

Here’s a table to demonstrate the differences between these business phone systems:

PRI SIP trunking Hosted VoIP
1. Upfront costs Medium-High High Low
2. Maintenance costs Medium-High Medium-High Low-High
3. Connectivity Physical Virtual Virtual
4. Service quality Low; calls may experience muffled or distant quality, frequency range is limited High; good bandwidth required, low bandwidth can lead to jitter, packet loss High; good bandwidth required for VoIP, low bandwidth can lead to jitter, packet loss
5. Scalability Low High; very scalable High; scalable
6. Mobility None; no routing ability Medium; calls can be transferred to predetermined locations Very high; can be used anywhere and through any device

Choosing the Right Phone System for Your Business

The phone system that is ideal for your business purposes depends on what you want to accomplish with it and what your budget can include. Speak with our experts today to see if VoIP or SIP trunking is a good fit for you!

6 Ways to Fix VoIP Jitter

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When conducting business calls, interruptions, low call quality, or missing audio can lead to miscommunication. Part of running a professional business is ensuring that your business calls, whether for queries or support, occur smoothly without any distortion or jumbled audio. Interruptions during calls can lead to losing valuable clients. One important element that affects VoIP business calls is jitter. In this post, we explain what leads to jitter and how to fix VoIP jitter in 6 useful ways.

Why You Need to Fix VoIP Jitter

In order to fix VoIP jitter, one must understand VoIP jitter and how it affects a business’ VoIP phone system. During VoIP business calls, voice messages are transformed from analog to digital signals and stored in data packets. For VoIP calls to connect two end-points successfully, data packets need to be transmitted effectively without delay or disturbance.

While these data packets move from one end-point to the next, the packets travel through different paths and may not take the same path. However, due to a variety of reasons — such as low internet speed, a low-quality router, and so on — the data packets may not be delivered at the same time. Instead, they may arrive at irregular intervals affecting VoIP call quality. Additionally, this can lead to missing or jumbled audio. This is known as ‘VoIP jitter.’ Jitter within business calls can lead to miscommunication and frustration for users. Here are 6 reliable ways to fix VoIP jitter:

1. Invest in a Powerful Router

When purchasing a router for your VoIP phone system, do your research and find one that is powerful and can handle your VoIP needs, especially the bandwidth capacity. Carefully review the product and see if it matches your needs. Study customer reviews and testimonials and look for complaints and potential issues.

2. Utilize an Ethernet Cable

Use a high-quality ethernet cable to connect your VoIP system to your router. This way, you will have a better connection and no interference from sources out of your control that can lead to jitter, latency, packet loss, and more. Additionally, if you already have an ethernet cable but are still experiencing jitter, then perhaps it’s time to upgrade your ethernet.

3. Subscribe to High-Speed Internet

Next, as is widely known, low internet connection speeds can affect the quality of your VoIP phone system. Low internet speeds lead to jitter, latency, and more. Make sure that your business has high-speed internet connection to ensure smooth connectivity.

4. Conduct Bandwidth Tests

Besides securing a high-speed internet connection, you also want to ensure that your bandwidth is strong enough to carry the weight of your VoIP phone system. Ask your ISP to test your bandwidth and then resolve jitter issues. You may even connect with your VoIP phone service provider for help in resolving VoIP jitter issues.

5. Consider Getting a Jitter Buffer

Another way to fix VoIP jitter is by using a jitter buffer, a device that intentionally delays an incoming data packet. By delaying an incoming packet, the receiver of the call will hear the voice message clearly and with very little distortion. This is because the jitter buffer will re-group delayed data packets and then play them together, steadily. Your data packets will be stored in the right sequence and played accurately and clearly.

6. Reduce Unnecessary Bandwidth Usage

Lastly, make it a practice to reduce unnecessary bandwidth usage, especially during office hours. Teach your staff to not use large amounts of bandwidth for non-work-related activities. This includes streaming videos or content from Netflix, etc. These services use large amounts of bandwidth and can lead to jitter during VoIP calls.

Convert More Customers with VoIP for Business

A business VoIP phone system can greatly improve the way your business communicates with its customers. Additionally, getting this service from a reliable VoIP provider can help improve VoIP call quality issues such as jitter, latency, and so on. Ready to upgrade your business phone system and get VoIP? Speak with our representatives today!

SIP Response Codes: A Complete Guide

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Learn about SIP response codes, how they function, and the different types of response codes available. Understanding SIP codes can help you identify issues within your communication system.

What are SIP Response Codes?

Session Initiation Protocol (SIP) is a signaling protocol used to facilitate and control communication sessions. As such, SIP lets users make and receive calls over the internet instead of traditional phone lines. This paves way for unified communications by enabling the transmission and sharing of voice, video, and other files.

A SIP session is based on a request/response transaction. Therefore, each session consists of a SIP request and at least one SIP response. Response codes indicate the status of the SIP request when making a connection between two or more parties.

How Do SIP Response Codes Work?

SIP responses use a 3-digit response code to outline or detail the status of a SIP request. For example, was the SIP request accepted, was it a bad request, and so on. These codes are divided into 6 broad categories, namely:

  1. Informational/Provisional
  2. Success
  3. Redirection
  4. Client error/Request failures
  5. Server error
  6. Global failure/error

These codes also contain a “reason phrase” which can be varied to provide additional information or in a different language.

Different Types of SIP Response Codes

So, what are the different types of SIP response codes and what do they indicate? Important abbreviations to be aware of:

  • User Agent Client (UAC) – initiates the requests
  • User Agent Server (UAS) – responds to the requests
  • Uniform Resource Identifier (URI) – a string of characters that unambiguously identify a particular resource

Here we will look at each response code in each category in detail:

1xx = Informational SIP Responses

1xx SIP response codes are sent at any time when a connection between two parties is being created. Common 1xx codes are:

100 – Trying: The request was received and an extended search or unspecified action is being performed.

180 – Ringing: The user agent has received an INVITE (SIP request code) and is alerting the user.

181 – Call is Being Forwarded: The call is being forwarded to another destination, receiver, endpoint.

182 – Queued: Indicates that the destination is temporarily unavailable and the server has placed the call in queue.

183 – Session Progress: Provides information about the progress of the call.

199 – Early Dialog Terminated: Indicates that an early dialogue has been terminated. Usually sent by the User Agent Server.

2xx = Success Responses

2xx codes indicate that the SIP request was received, understood, and accepted. Common 2xx codes are:

200 – OK: Indicates that the request was successful.

202 – Accepted: Indicates that UAS has received and accepted the request, but it has not been authorized or processed by the server yet.

204 – No Notification: Indicates that the request was successful. However, no response will be received.

3xx = Redirection Responses

3xx response codes inform the UAC about redirections and further action is needed to complete the request or reach the UAS.

300 – Multiple Choices: The request address returned several choices with different locations. The UA can select one of several options of endpoints to redirect the request.

301 – Moved Permanently: The user is no longer at the address used in the request. The original request URI is no longer valid. A new address will be provided in the Contact header field. This address should be saved and used in the future.

302 – Moved Temporarily: A new address will be provided in the Contact header field. The UAC should try the new address. This address should not be saved for the future.

305 – Use Proxy: To access the destination and address, a proxy is required. The proxy will be displayed in the Contact field.

380 – Alternative Service: The call failed, but alternatives are noted in the message body.

4xx = Request Failures/Client Error

4xx response codes indicate that the message was not processed due to an error. The request may include bad syntax and therefore cannot be fulfilled at this server

400 – Bad Request: Indicates that the request could not be understood.

401 – Not Authorized/Unauthorized: Indicates that the request requires user authentication.

403 – Forbidden: Indicates that the server is refusing to fulfill the request, even though it has understood it.

404 – Not Found: The user does not exist in that particular domain.

405 – Method Not Allowed: The method specified in the Request-Line is understood, however, it is not allowed.

406 – Not Acceptable: The resource can only generate responses with unacceptable content.

407 – Proxy Authentication Required: Similar to the 401 code, the request requires user authentication.

408 – Request Timeout: The server couldn’t find the user within a suitable time frame.

409 – Conflict: User already registered (deprecated).

410 – Gone: The user is not available here anymore.

411 – Length Required: The server needs a valid content length before accepting the request.

412 – Conditional Request Failed: The given precondition has not been met.

413 – Request Entity Too Large: Indicates that the request message body is too large.

414 – Request URI Too Long: The server refuses to accept the request. This is because the request URI is longer than the server can interpret or understand.

415 – Unsupported Media Type: Requested message body is in a format that is not supported by the server.

416 – Unsupported URI Scheme: The request URI is unknown to the server or not supported by the server.

417 – Unknown Resource-Priority: Indicates that a resource-priority option tag was present, but without a Resource-Priority header.

420 – Bad Extension: Bad SIP Extension was used. The SIP extension is not understood by the server.

421 – Extension Required: The server requires a specific SIP extension that is not listed in the supported header.

422 – Session Interval Too Small: The request contains a Session-Expires header field with a duration or interval that is too small or below the minimum.

423 – Interval Too Brief: Similar to 422, the expiration time of the resource is too short.

424 – Bad Location Information: The request’s location content was unsatisfactory or “bad.”

428 – Use Identity Header: An Identity header field is required by the server policy and one has not been provided.

429 – Provide Referrer Identity: The server has not received a valid Referred-By token on the request.

430 – Flow Failed: A specific “flow” that was sent to a user agent has failed. However, other flows may succeed.

433 – Anonymity Disallowed: The request was rejected since it was anonymous.

436 – Bad Identity Info: The request has an Identity-Info header filed and the URI contained cannot be identified.

437 – Unsupported Certificate: The server could not validate a certificate for the domain that signed or sent out the request.

438 – Invalid Identity Header: Server obtained a valid certificate used to sign a request. However, the server could not verify the signature.

439 – First Hop Lacks Outbound Support: The first outbound proxy doesn’t support the “outbound” feature.

440 – Max-Breadth Exceeded: A client that received a 440 response can interpret that its request did not reach all possible destinations.

469 – Bad Info Package: A 469 response indicates that the receiver is not willing to accept this Info Package.

470 – Consent Needed: The source of the request did not have the recipient’s permission to make such a request.

480 – Temporarily Unavailable: The recipient is currently unavailable.

481 – Call/Transaction Does Not Exist: The server received a request that does not match any dialogue or transaction.

482 – Loop Detected: Server has detected a loop.

483 – Too Many Hops: Max-Forwards header has reached the value ‘0.’

484 – Address Incomplete: The requested URI is incomplete.

485 – Ambiguous: The request-URI is ambiguous.

486 – Busy Here: The recipient is busy.

487 – Request Terminated: Request has terminated or canceled.

488 – Not Acceptable Here: Parts of the session description of the request URI are not acceptable.

489 – Bad Event: The server could not understand an event package specified in an Event header field.

491 – Request Pending: Server has some pending requests from the same dialogue.

493 – Undecipherable: The request contains an encrypted MIME body, which the recipient can not decrypt.

494 – Security Agreement Required: The server has received a request that needs a negotiated security agreement.

5xx = Server Errors

5xx response codes indicate that there’s an issue with the server and it has, therefore, failed to fulfill a valid request.

500 – Server Internal Error: The request could not be fulfilled due to some unexpected condition.

501 – Not Implemented: The SIP request method is not implemented here.

502 – Bad Gateway: An invalid response was received from a downstream server while trying to fulfill a request.

503 – Service Unavailable: The server is in maintenance or temporarily overloaded. Therefore, cannot process the request.

504 – Server Time-out: The server tried to access another server while attempting to process a request. However, there was no timely response.

505 – Version Not Supported: The SIP protocol version in the request is not supported by the server.

513 – Message Too Large: The length of the request message is longer than the server can process.

555 – Push Notification Service Not Supported: The server does not support the push notification specified in the SIP URI parameter.

580 – Precondition Failure: The server is unable or unwilling to meet the constraints specified in the request.

6xx = Global Failures/ Global Error

The request cannot be completed at any server.

600 – Busy Everywhere: All possible destinations are busy.

603 – Decline: Destination cannot participate in the call and there are no alternative destinations.

604 – Does Not Exist Anywhere: The requested user does not exist anywhere.

606 – Not Acceptable: The user’s agent was contacted successfully. However, certain aspects of the session description are not acceptable.

607 – Unwanted: The call is unwanted by the recipient. Future attempts are likely to be similarly rejected.

Buy Quality SIP Trunks from United World Telecom

Buy SIP trunks from us and improve the way your business communicates with advanced features, high voice quality, and competitive rates. Sign up on our website or speak with our specialists to learn more!

Related: SIP Trunk Pricing Breakdown (2020)

8 Powerful Applications Built Using WebRTC

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Want to know how powerful Web Real-Time Communications (WebRTC) can be for an app or browser client? Here are 8 great applications built using WebRTC that are currently being used by millions around the globe.

WebRTC Applications: 8 Powerful Examples

First, what is WebRTC? Web Real-Time Communication is a communication framework that is open-source for web browsers and phones. It is a free project that gives websites real-time communications capabilities, making audio and video communication possible. WebRTC applications can be accessed through most web browsers like Chrome, Mozilla, Safari, Microsoft Edge, etc. Additionally, they can be accessed on Android, Samsung, and iOS devices.

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Let’s look at 8 powerful applications built using WebRTC and how they work.

1. Google Hangouts, Google Meet, Google Duo

Since 2011, Google has been using Web Real-Time Communications and has developed multiple communication apps for personal and business use. These apps include: Google Hangouts, Google Meet, and Google Duo.

Google Hangouts was the first to offer voice and video calls as well as online messaging and SMS. Google Meet developed as an extension to Google Hangouts as a premium video conferencing tool. It supports more users as well as speech-to-text transcription. Google launched the video calling app, Duo, in 2016 for Android and iOS users. Its use of WebRTC has led to peer-to-peer connectivity and end-to-end encryption, making it secure and reliable.

2. Facebook Messenger

Facebook’s mobile app and web client (accessible through a web browser) are both powered by WebRTC. By using Web Real-Time Communications, Messenger has brought voice and video calls to its users, and more recently, allows for co-broadcasting via Facebook Live. Additionally, Facebook has also incorporated WebRTC in VR Chat for video calls in Oculus, Workplace by Facebook, and IG Live Video Chat.

3. WhatsApp

Started as a simple messaging service, WhatsApp has grown into a global messaging platform connecting users from around the globe quickly. WhatsApp’s Android and iOS apps heavily use WebRTC as well as utilize SIP calling for fast and reliable virtual communication.

Since its inception, users can now send voice notes as well as make voice and video calls over the internet. Additionally, more recently, WhatsApp became web-accessible through its web client. Users log into web browsers and use a QR code to access their messages through the browser.

4. Amazon Chime

Amazon, like the many apps and services it has offered over the years, also has a video conferencing tool called Chime. Chime is an internal video conferencing tool that uses Web Real-Time Communications in its services including Kinesis Video Streams, and Alexa’s smart home integration (cameras and doorbells). It seems that these applications have integrated WebRTC with existing communication technology such as VoIP and SIP systems.

5. Houseparty

Houseparty, the app of 2020, is a group video chat that became popular during Covid-19 lockdowns. The pandemic led to social distancing and a desperate desire for social interactions. As such, people started to look for online services that would help connect them with their loved ones. Enter: Houseparty. Using WebRTC, Houseparty provides real-time group communication and peer-to-peer video chat. Even though the rise of this company can be attributed to the pandemic, its services and popularity are here to stay.

6. GoToMeeting

GoToMeeting had used various VoIP technology and WebRTC functions in their web client video conferencing. Most of their customers and users have largely utilized the desktop client (non-WebRTC). However, growing popularity with the easy-to-use web client is drawing more users to use the browser tool.

7. Discord

Originally developed for the online gaming community, Discord combines Web Real-Time Communications and VoIP to bring voice calls and in-app messaging to its users. Discord’s engineering blog details how they have used WebRTC to serve more than two million users concurrently. They have over 87 million registered users and about 14 million active users daily.

8. Snapchat

A social media favorite, Snapchat is an app used by millions among the younger generation. Originally a platform for sharing ‘snaps’ of everyday life, the app now also boasts a video chat feature. This feature comes after Snapchat acquired AddLive, a WebRTC company that provided voice and video chat to the app.

What Can You Do with WebRTC?

As you can see, companies have used Web Real-Time Communication to develop stronger apps and browser clients. And as a result, they have made communication across boundaries quicker and more reliable. Your business can also improve its overall communication system and provide customers with better communication with these applications. To learn more, speak with our experts today!

Understanding Voice Over IP Jitter, Latency, and Packet Loss

The key to good VoIP call quality depends on a few factors such as jitter, latency, and packet loss. We discuss these elements below so you can ensure your business has strong and reliable VoIP quality for customer calls.

Understanding VoIP Call Quality: The Basics

Voice over IP (VoIP) calls use the internet to transmit voice or data packets from one user to their destination. On VoIP calls, your voice is transformed from analog to digital signals in data packets and is sent to your destination. Upon arrival, these packets are converted back to analog and the audio is heard. Data packets generally contain about 20 milliseconds of audio and this whole process occurs at lightning speed.

And while this process seems simple and straightforward, there are a few factors that can affect the quality of the call, interrupting it. Good VoIP call quality depends on keeping the following elements to a minimum:

  • Jitter
  • Latency
  • Packet loss

Let’s look at these issues more closely and ways to troubleshoot them.

Voice Over IP Jitter

For a VoIP or SIP call to take place successfully, data packets must be transmitted from one user to their destination. And these data packets travel through different paths before they reach the destination. As such, all data packets may not take the same path or time to arrive.

VoIP jitter refers to the data packets being delivered to the destination at irregular intervals instead of being delivered at the same time. In other words, one packet is delivered after the rest of the packet. This can lead to low VoIP call quality with missing or jumbled audio.

How to fix network jitter?

Generally speaking, 30 milliseconds (or less) jitter is acceptable. However, more than that can lead to serious call quality issues, affecting your calls and customer care efforts. And so, to fix jitter issues, you must first check your network and ensure you have a good internet connection.

Another way to fix network jitter issues is by using a jitter buffer. This is a space where packets are collected and stored. Then, they are sent out at regular intervals ensuring they move in the right order.

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VoIP Latency

Voice over IP latency refers to lag or delay within the call. More specifically, it’s the delayed time between a caller speaking and the receiver hearing the audio. This lag or delay can lead to speakers talking over each other or echoes in the middle of the call.

It is also important to note that international calls may experience more latency than domestic or local calls. And while it is not desirable, users generally tend to accept latency in long-distance calls more than local ones.

How to fix this issue?

Latency does not necessarily affect VoIP call quality. However, it does make the caller experience less desirable, giving way to frustration and miscommunication.

Most of the time, latency is a result of network congestion, which also contributes to jitter. To combat this, you should prioritize voice over IP data ahead of other data transmitted across your network. And a high-quality VoIP router can help with this as well as other issues that may crop up within a VoIP phone system.

Voice Over IP Packet Loss

Understanding packet loss is pretty straightforward. It refers to data packets lost during transmission from one user to their destination. Packet loss occurs when:

  • Data packets are lost and never arrive at the destination
  • Packets arrive late and are discarded as a result
  • Packets contain errors and are discarded
  • High data packet loss which results in low VoIP call quality or missing pieces of audio.

When data packets go missing, communication between two parties is incomplete or unclear. Troubleshooting this issue is similar to fixing network jitter and latency: check your network. Congested networks where multiple and large files are downloaded or uploaded or transferred can lead to packet loss. Therefore, to ensure low to no packet loss:

  1. Make sure you have enough bandwidth.
  2. Minimize network congestion (don’t stream videos or download music or send large email attachments).

Get a Reliable VoIP Provider

To ensure you do not suffer through these issues, it is important to find a VoIP number provider that can handle your voice over IP traffic. Learn more about our VoIP service by speaking with one of our experts today. Call us at 1 (877) 898 8646 or chat with us online for more information!

What are the Different Types of Contact Centers?

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Can’t decide which of the different types of contact centers is right for your business communication needs? Here we highlight 7 types of call centers to help you understand which is ideal for your purposes.

Understanding the 7 Types of Contact Centers

There are a few different types of contact centers that exist with different focuses and purposes. This ranges from centers that have it all or centers that focus on incoming or outgoing calls, those that use specific virtual call center software to centers offering multichannel communication options, and so on. You can also outsource your communication needs to some centers. On the other hand, other types of call center software can give your business access to contact center tools to use in-house.

However, which type of call center or call center software that will work for you depends on your specific needs. So, let’s look at the 7 types of contact centers:

  • Call centers
  • Contact centers
  • Inbound centers
  • Outbound centers
  • On-premise centers
  • Cloud-based or virtual centers
  • Multichannel or omnichannel centers

Let’s look at each of these types of contact centers individually.

1. Call Centers

The terms call center and contact center are often used interchangeably; however, there are a few differences between them. A call center, for instance, is a centralized center where reps answer incoming calls from potential and current customers of various businesses. Some call centers handle only incoming or outgoing calls while others handle both, also called ‘blended’ centers. Additionally, a business can have an in-house or on-premise call center or they can outsource their needs to a company specializing in call center services.

2. Contact Centers

Contact centers are similar to call centers except that they are more multichannel or omnichannel. This means that along with receiving calls, these centers also offer email, SMS, live chat, and social media communication channels. Call centers usually stick to phone conversations only while contact centers offer more channels and modes of contact.

3. Inbound Centers

Inbound contact centers focus primarily on incoming calls. This means that they have trained agents and reps to answer calls and provide sales or customer support services. Most inbound contact centers are generally customer service-oriented. Customers generally call a business for a few reasons:

  1. Inquire about a product or service
  2. Ask for technical support
  3. Receive assistance with a purchased product or service

Usually, an IVR system answers the call and interacts with the caller by offering menu options. Then, it proceeds to help the caller via pre-recorded messages or by transferring the caller to the right department.

The goal of inbound centers is to handle customer calls and concerns quickly and efficiently. This means answering and resolving calls professionally. This helps retain more customers by increasing customer satisfaction.

4. Outbound Centers

Outbound contact centers do the opposite of inbound centers. That is, they focus primarily on outgoing calls and lead generation. These contact centers call lists of potential clients or leads in an attempt to make new sales. Outbound centers use customer relationship management (CRM) systems to keep track of contacts, leads, and calls. Some outbound centers offer additional outbound calling services such as fundraising, collecting customer feedback and surveys, outreach efforts, and more.

5. On-premise Centers

Many types of contact centers work on-premise or in-house. This means that the call center works within your office and all the hardware and software are operated and managed by your in-house IT team.

On-premise centers are known for their high level of data security and therefore tend to be more reliable and have better call quality. Additionally, you will have total control over your communication system and you can use it according to your needs. However, running your contact center on-premise means that your business will be in charge of purchasing and maintaining hardware, hiring a highly-skilled IT team, and paying other upgrade costs. All of this can lead to higher costs for your business.

And so when deciding whether you need an on-premise center, consider this: do you have the budget, infrastructure, and IT team to handle the system in-house?

6. Cloud-Based or Virtual Call Centers

Cloud call centers are an alternative to on-premise centers. They work virtually and are hosted by your provider. In other words, your provider runs and manages your call center software while you simply use the service.

A cloud-based contact center gives you less control. However, your business is not responsible for any hardware or maintenance costs, which can save substantially on expenses. Your teams can access the software from any location or device as long as they have a good internet connection. Most businesses that manage remote teams use virtual call center software to provide their teams with the right tools needed for excellent customer service.

7. Multichannel or Omnichannel Centers

Multichannel and omnichannel centers are one of the most effective types of contact centers that offer not just voice but other communication channels as well. This includes video, email, SMS, live chat, and social media engagement. One thing to note is that multichannel centers may not offer all communication channels while omnichannel centers do.

Having multichannel support can help your business reach more customers across different channels. Interested customers who do not prefer phone conversations can use other means to connect with your business, which helps you increase your customer base.

Choosing the Right Call Center Software for Your Business

Businesses of every type can use call center or contact center software to improve the way they interact with their customers. You can enhance caller experience, increase customer satisfaction, and in turn, improve your business’ overall sales. Reach out to us today to learn more!

What is WebRTC and What is it Used for?

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WebRTC is a communication program that enables real-time and effective communication on different devices. And by doing so, it gives users — individuals and businesses — the tools they need for high-quality voice and video calling.

WebRTC Explained

WebRTC or Web Real-Time Communication is a free, open-source project for web browsers and web phone apps. More specifically, it is an HTML5 specification that adds real-time communication to browsers by working inside web pages.

In other words, this program provides browsers and apps with RTC through APIs making audio and video communication possible without the need for plugins. And since 2011, this project has grown popular with video calling as an excellent and desirable addition.

You can get Web Real-Time Communication for most modern browsers including Chrome, Mozilla Firefox, Safari, and Microsoft Edge.

How Does WebRTC Work?

Before the age of Web Real-Time Communication, creators needed C/C++ to build programs for voice and video calling. This meant long development cycles and higher costs. However, WebRTC replaces C/C++ with an API which can be used from inside a browser. This resulted in the cost-effective and easy development and integration of unified communications in real-time.

The purpose of Web Real-Time Communication is to make live, real-time interactions possible. With WebRTC, users gain access to various devices or parts of your device. For example, you can access and use your laptop, phone, or computer’s microphone, camera, etc. Furthermore, you can even capture or record the screen and share your screen remotely.

4 Benefits of WebRTC

Web Real-Time Communication is used widely because of the many benefits of such a feature. Here is a list of the top reasons to use WebRTC for unified communications:

  • It is an open-source program that is free for personal and business use.
  • It is available for all modern and commonly used browsers so you can access it no matter what operating system you use.
  • WebRTC is not only available for browsers but also mobile and smartphone apps.
  • Along with voice and video calling, you can use Web Real-Time Communications to create calling groups, record VoIP calls, and more.

Improve Business Communications

Web Real-Time Communication enables users to access voice and video calling from any device, thereby, making business communication easy and accessible. To learn more, speak with one of our experts today!

What Is ISDN? The Integrated Services Digital Network Explained

Want to establish an effective and easy-to-use business phone system? Here is an in-depth post about Integrated Services Digital Network (ISDN), how it works, major benefits for business uses, and viable alternatives.

Integrated Services Digital Network Explained

Integrated services digital network or ISDN is a telephone network system. ISDN uses circuit switches to send and receive voice and data through a digital copper line. This phone system was created to replace old landline technology and make it more digital. By doing so, ISDN provides better speed and transmission quality.

How Does ISDN Work?

When users do not have access to a Digital Subscriber Line (DSL) or cable modem connections, then ISDN is the best solution for high-speed internet. To set up ISDN, you will have to work with your ISP and have a working POTS line and phone number.

ISDN vs PSTN vs DSL

When discussing phone-internet services, you will come across a variety of terms such as ISDN, PSTN, POTS, DSL, etc. So, let’s discuss and define some related terms and how they work.

A public switched telephone network is a phone network that uses fixed lines with copper wires to connect users. While the phone network service is almost entirely digital now, the signal transmitted from one user to the next is still analog. Common PSTN features include:

  • Line hunting
  • Voicemail storage
  • Fax line
  • Custom messages
  • Caller ID

ISDN works as an integrated service. In other words, it is used to transmit voice, video, data, and other unified communications between users. By adopting ISDN, users (businesses) do not have to purchase, install, and manage multiple PSTNs. Common ISDN features include:

  • Multiple numbers in a block
  • Extensions and DID numbers assigned to individuals or departments
  • Line hunting

DSL takes ISDN to the next level by digitizing the system. And by doing so, it transmits data quickly and with better quality. But how does it do so? ISDN is a dial-up service that works through one line whereas DSL is always connected and does not need to be dialed to work. This results in DSL sending packets at a speed of up to 100Mbps, while ISDN slows down after 128Kbps.

4 Major Benefits of ISDN

So, why should businesses opt for ISDN and what benefits can they expect from this service? Here are four major advantages of an ISDN connection:

  • ISDN offers better quality and faster data transmission. By doing so, it provides a higher data transfer rate.
  • Allows for multiple data transmissions simultaneously.
  • Users can access multiple digital services through the same copper wire.
  • Users can operate multiple devices (phones, computers, fax machines, credit card readers, etc.) through the same line.

what is isdn

ISDN Alternative: VoIP

While ISDN seems like an effective business communication service, it is not the most cost-effective option out there. A popular alternative to ISDN is voice over IP. A VoIP phone system converts audio signals into digital signals and transmits them from one user to the next through an internet connection.

By doing so, VoIP lets users make and receive calls without the need of a physical telephone line. Additionally, users can operate this service from any location and through any device of their choosing, making business communications more reliable and accessible. Benefits of using a VoIP phone system include:

1. Cost-effectiveness: A landline phone system costs about $50 for a line that can only call local and domestic numbers. VoIP, on the other hand, gives users access to local numbers and international toll free numbers, and more for low, bundled prices.

2. Features: Gain access to a variety of advanced virtual communication tools such as:

    1. International call forwarding
    2. Extensions for different departments
    3. Creative and effective call management tools
    4. Outbound calling with a dynamic caller ID
    5. Call recording
    6. Call analytics

3. Accessibility and Flexibility: Be available and accessible from any location and device. Make remote working possible and accountable by using a reliable phone system.

4. Scalability: Grow and expand your business phone system as and when needed. Installing new VoIP lines does not mean installing a new connection but acts more like getting a new username which can quickly be connected to a device.

Want to Learn More?

Our telecom experts can help you understand the different types of phone communication services and features available. Furthermore, you can identify which services and tools can boost your specific business’ image and customer relations. Call us today or chat with us online to learn more about the options ahead!

5 Unified Communications Trends You Need to Know in 2025

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Every year, businesses need to check themselves and see how they stack up against not just competition, but market trends. This is where unified communications trends of 2025 come in. How can businesses continue to improve their communication system and, by extension, maintain productivity and manage remote teams?

Unified Communications Trends: 2025

The unified communications trends for 2025 seem to focus not only on increased productivity, advanced technology, and better customer service but also on ways to improve employees’ experience and team collaboration. Read on for the top UC trends for 2025 that businesses should keep in mind.

1. Adoption of UCaaS

One of the top UC trends is the growing adoption of UCaaS. Large companies and enterprises are investing in unified communications as a service as a business communication system. This is because UCaaS consolidates major communications channels and platforms such as voice, video, SMS, and email in the cloud. And by combining all of these applications and channels in one place, UCaaS boasts a cost-effective and organized business phone system.

2. Rising Presence of AI

As can be seen with most industries, artificial intelligence (AI) is slowly expanding and making its way in major processes. Telecom and business communication is no stranger to AI. However, a growing unified communications trend sees AI becoming a more common tool. Think: virtual chatbots, virtual assistants, integrated services, and more. The implementation of AI resonates with the goal of using technology to improve productivity and collaboration and achieve faster and more accurate results.

3. More Cloud Solutions

Cloud communication solutions continue to be on the list of UC trends. And this is simply because cloud computing and storage have been in great demand and used by almost every business. According to a 2019 study conducted by Nemertes, about 67% of companies are currently using some part of their UC cloud solution while a third of those companies are using their UC cloud solution to the fullest extent. Because of this, UC providers are looking at partnerships, affiliations, and integrations that enable users to take advantage of more comprehensive cloud solutions.

4. Rise of Global SIP

As businesses across the world are renewing and improving their global communication systems, it is important for your company to stay ahead of the game. Global SIP gives multinational businesses the ability to expand with ease and reduce communication-related costs. With SIP trunking from VoIP providers, your business can place focus on voice network infrastructure, offer multichannel and multimedia support, and have access to centralized cloud communications.

5. Focus on EX as well as CX

While most UC trends are focused on improving customer experience (CX), in 2025, focus has shifted to employee experience (EX). Improved EX directly leads to enhanced CX, which at the end of the day should be every business’ goal. Unified communications can help employees become more efficient; efficient employees reduce costs and increase customer satisfaction.

Switching to UC for Business

Unified communications can vastly improve your business’ productivity while ensuring it stays in the game with advanced telecom technology. Find out how United World Telecom can help your business; speak with one of our experts today!

Why VoIP is a Good Fit for Distributed Teams

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Need a business communication solution for distributed teams and remote working? Voice over IP or a VoIP phone system can be a great fit for your distributed teams. It can help to ensure your teams stay connected and productive even when they work remotely.

4 Ways VoIP Can Benefit Your Distributed Teams

A VoIP phone system ensures business connectivity over the internet, enabling users to connect from any location and any device. This makes it a good fit for distributed teams and remote workers. Read on for the top 4 benefits of using VoIP to manage remote groups.

1. Employees Can Work from Any Location

Whether your area is hit by a pandemic or you plan on hiring employees from different states and countries, the best part about having VoIP for business is that your employees can stay connected and continue working from wherever they are. All they need is to connect to your VoIP system through a reliable internet connection. Then, they can place and answer important business calls, make sales calls, and offer customer support uninterrupted.

2. Remain Accessible through Multiple Devices

Secondly, employees are not limited to the phones within an office. Employees in distributed teams can work through any device — VoIP phones, computers, laptops, smartphones, tablets, etc. This ensures that they remain accessible even when they are not in the office or if a device fails.

3. Implement a BYOD Policy

Bring your own device or BYOD work policies are growing more and more popular. These policies make it possible for employees to use their own devices instead of devices issued by the company. This means that employees can use devices that are suitable, familiar, and comfortable to use. Additionally, they do not need to learn how to use a new device and can use their own apps that help with productivity. It also helps businesses reduce equipment costs as they do not need to provide employees with phones, computers, laptops, etc., to get their work done. All in all, a BYOD policy can be a win-win situation for both employees and employers.

4. Reduce Communication-Related Costs

With all of the above taken into account, you’ll notice that VoIP can help your business reduce communication costs. For one, you will have a predictable bill each month. Two, you don’t have to worry about equipment or maintenance costs. And that also means you do not have to bother with IT costs. When businesses employ remote workers, having the ability to maintain communication without needing to purchase additional equipment or services can really help keep expenses to a minimum.

How to Set Up VoIP for Remote Teams

Setting up VoIP for distributed teams is quick and easy. Once you have the service, you will need to map out how you want your teams to use the service and its benefits. Here are some key points to keep in mind:

1. Map Out Your Phone System

The first step to setting up a VoIP phone system for your remote workers is to determine how you’ll want calls to be handled. For instance, how many numbers will you be using, and for which departments? Map out how calls will move through the system and how calls will be forwarded or routed. Determine your forwarding needs and settings. For example, you can forward calls to different people at different times of the day or simultaneously to every worker, and so on.

2. Decide What Features to Use

Next, you need to decide how you want to use the features that come with your VoIP service. For instance, you should set up a custom greeting and voicemail message. You will also want to design an IVR with menu options for callers to navigate through. You can play around with other features to see which ones work best for you and your teams.

3. Equip Your Agents

Next, prepare your remote team and equip them to handle business calls. Here too you can use VoIP features to ensure your agents have what they need to do their job effectively. You may even consider creating a support portal or knowledge base that walks them through common issues, troubleshooting help, and so on. Lastly, train them to use the service as well as to be effective agents and salespeople. Once all of this is done, it’s time to start working!

4. Access In-Network Calling for Team Collaboration

VoIP allows for in-network calling which is essential to remote team collaboration and manager-employee interaction. This is especially true for distributed teams and remote workers. Calling someone from your team should be without hurdles and interruptions. And in-network calling makes team communication easy.

5. Make Outgoing Calls from Own Devices with Business Numbers

An outbound calling service from your VoIP provider means that users can make outgoing calls from any device while still displaying a dynamic caller ID. And so, receivers of these calls see your business number even if a remote worker is calling from their laptop. This keeps business calls and remote working professional.

6. Measure Results with Call Records

Lastly, measure results and important KPIs to see how your teams are doing on a regular basis. And you can do this through studying call detail records and call activity. Watch for top KPIs such as first call resolution, average time in queue, average handle time, response time, and so on.

Think VoIP is a Good Fit for Your Remote Teams?

Get a VoIP phone system with United World Telecom and improve distributed teams’ business communication. Give your remote workers the tools they need to work effectively and produce the desired results. For more information, call us today or chat with our experts!