VoIP

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

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

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

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

Building a Business Phone System: VoIP versus UCaaS

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

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

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

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

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

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

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

Let’s look at these solutions more closely.

What is VoIP and How Does it Work?

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

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

What can you do with VoIP?

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

Benefits of VoIP:

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

What is UCaaS and How Does it Work?

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

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

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

What can you do with UCaaS?

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

Benefits of UCaaS:

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

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

Which Communication Solution Should I Choose?

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

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

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

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

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

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

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

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

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

Reliability of VoIP

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

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

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

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

Client-Side Factors

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

Bandwidth requirements:

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

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

Router optimization:

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

Backup power and internet sources:

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

Equipment:

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

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

Provider-Side Factors

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

So, what makes a VoIP provider reliable?

99.999% uptime:

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

Global and local servers:

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

Dedicated account management:

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

24/7 technical support:

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

Regular monitoring:

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

Is VoIP More Reliable than POTS?

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

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

Here’s a brief overview of PSTN versus VoIP:

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

Source: Statista

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

Source: Bullseye Telecom

When is VoIP the Better Choice?

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

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

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

How to Choose a Reliable VoIP Service Provider

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

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

Get VoIP with United World Telecom

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

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

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

What is One-Way Audio?

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

Why Does It Occur?

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

The most common causes include:

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

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

How to Troubleshoot One-Way Audio: 4 Tips

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

1. Check Your Equipment

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

Equipment Issue Solution

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

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

2. Review Your Router’s Settings

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

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

Firewalls / NAT Blocking Voice Data Solution

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

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

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

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

3. Ensure Devices Share Common Codecs

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

Incompatible Codec Solution

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

4. Reconfigure Routes

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

Routing Misconfiguration Solution

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

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

Solve Call Quality Issues with a Reliable Provider

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

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

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

Voice over IP is the new standard when it comes to business telephony.

In VoIP, voice is converted into data packets and sent over the internet using communications protocols. Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) are both commonly used protocols for data transfer over the internet.

But how do you decide which is better for your business VoIP system? Here we compare TCP versus UDP, so you understand how these protocols affect VoIP calls.

Is TCP or UDP Ideal for VoIP?

The short answer is that UDP provides a better experience for VoIP users. Using UDP will help minimize network jitter and latency to ensure strong connections for your business phone calls.

Why Do TCP and UDP matter for VoIP?

Well, VoIP sends data packets over the internet.

These data packets consist of a header, which includes information about where the packet comes from and where it should go. They also have a payload, which is the actual data being transported by the packet.

TCP and UDP are just two different types of data packets. Both carry a payload and use internet protocols, but each works slightly differently.

Why is UDP Better Than TCP for VoIP?

Well, VoIP traffic is fast-flowing. A typical data packet contains 10 to 30 milliseconds of audio. Essentially, many, many data packets are sent one after the other. If a few packets are lost here and there, your conversation will probably remain unaffected.

Here’s an example:

Picture a well-trafficked highway; if one of the cars on the highway breaks down, you wouldn’t stop the flow of traffic just to wait for that one car. It’s better to just maintain the flow of traffic.

In the case of VoIP, TCP tries to recover lost packets, causing retransmission delays which stop the flow of voice traffic. UDP lets voice traffic continue to flow.

If you wait for and try to recover lost packets, the quality of your VoIP calls is going to be very low.

Let’s look at TCP and UDP in more detail.

UDP – User Datagram Protocol

This protocol is one of the core members of the IP suite.

UDP is generally better for real-time and high-performance applications like VoIP, which can’t afford retransmission delays. With UDP, there are no retransmission delays.

Here are some of the factors that make UDP ideal for VoIP:

• Connectionless — Avoids the overhead associated with establishing a 3-way handshake.
• No retransmission delays — If packets are lost, TCP will try to retransmit and recover lost packets, causing delays. This is not the case with UDP.
• Speed — UDP is faster, simpler, and more efficient than TCP.

However, UDP also has some downsides compared to TCP:

• No guaranteed ordering of packets — VoIP audio might be transmitted in a slightly different order than how it was sent.
• No protection against duplicate packets.

In some cases, these disadvantages can lead to jumbled audio.

However, UDP is still always recommended for VoIP calls.

TCP – Transmission Control Protocol

This protocol is one of the most commonly used protocols within the IP suite because it ensures the integrity of end-to-end data delivery over a network.

Here is a quick overview of the Transmission Control Protocol:

• Connection-oriented
• Good failure recovery
• High error rate handling and error-checking
• Sequencing – packets arrive in order at the receiver
• Relatively slower than UDP.

TCP is connection-oriented, which means that a connection must be established before data transmission takes place. The connection must also be closed after transmitting the voice packets. This comes with high overhead.

It also provides extensive error-checking mechanisms. If packets are lost, TCP will try to recover and retransmit those packets.

Your VoIP lines will become garbled due to packets trying to recover themselves, making it difficult to make sense of conversations.

TCP is ideal for applications like peer-to-peer file transfer, secure shell, and Internet Message Access Protocol (IMAP), for example. But, it’s actually harmful to low-latency connections like VoIP.

TCP versus UDP: What Do VoIP Experts Recommend?

As a network administrator or VoIP engineer, you’ll want to remember that UDP is ideal for VoIP and SIP connections.

Using UDP will help improve call quality significantly. It will also help with device registration. If you’re trying to register lots of VoIP devices on your network, TCP will slow down your servers.

To learn more about using the differences between TCP and UDP and which is better for VoIP, speak with one of our telecom experts.

Voice over IP or VoIP has been a vital business communications tool for many years.

While the technology is still developing, a large percentage of organizations around the world have already adopted it. VoIP has many benefits, including cost savings, scalability, and global accessibility.

However, some countries have banned VoIP, including the UAE and other Middle Eastern countries. Why? Let’s take a look.

Which Countries Have Banned VoIP?

Currently, there are around 20 countries that regulate or block VoIP calls. Here is a list of those countries, along with more details about VoIP restrictions in each country.

Why is VoIP Banned in Some Countries?

Countries around the world block or restrict VoIP calls for several reasons.

The first reason to block VoIP calling is to preserve government profits. Various state-owned telecom monopolies worldwide — especially in the Middle East — rely on international calling and voice communications as a major source of revenue. Large revenue streams would dry up if these countries allowed VoIP.

Security and privacy are additional reasons for countries blocking VoIP. Some countries claim that their VoIP bans serve to protect society from negative influences that could harm the public interest. Others block the technology for security reasons during times of political turmoil and to prevent cheating.

Reasons VoIP is Blocked in Dubai and Kuwait

UAE is an expat-majority country. Many people in the country want to call their loved ones back home and thus require international calling services. However, the country has two government-owned monopolies – Du and Etisalat. Both operators take measures to ensure VoIP does not endanger their revenues.

Kuwait has a similar situation where local authorities look to preserve their revenues derived from voice communications. If they allowed VoIP in Kuwait, they would lose a significant source of revenue.

How are VoIP Bans Affecting People?

VoIP bans are nothing new. In fact, they started years back in 1999, around the time the technology started becoming popular. The effects of these bans can be significant.

For example, many people living abroad cannot communicate with their families back home. The exorbitant international calling rates imposed by monopolies prevent people from calling their families.

As a result, people are growing more frustrated and sharing those frustrations online.

Looking Forward

Local and regional telecom operators want to preserve their profits. It’s understandable. But residents in many countries have called to end VoIP bans.

So, VoIP laws might change around the world. There have been ongoing negotiations between big tech companies and some of the countries blocking VoIP.

For example, Microsoft and Apple are negotiating with countries to repeal their bans on Skype and FaceTime. If these negotiations go well, perhaps the countries will also start to allow businesses to use VoIP. But for now, the technology remains banned in some countries.

VoIP Alternatives for Blocked Countries

You can use three major alternatives to VoIP in countries where restrictions apply. These three methods — VPN, local access numbers, and international callback — all offer significant cost savings compared to using services from telecom monopolies.

However, governments and ISPs in those countries know of the methods used to bypass their restrictions. While these VoIP alternatives may achieve their purpose, authorities may consider them illegal.

So, even though an alternative may work for a given time, it may soon incur restrictions or legal repercussions.

1. VPN Tunnel

A VPN tunnel is the first VoIP alternative for countries with VoIP restrictions. Users will simply connect to a VPN that encrypts their traffic and hides their IP addresses. They’ll be able to access VoIP software to make and receive calls.

2. Local Access Number

The second major VoIP alternative is using a local access number to dial out. Your service provider would offer you a local or freephone number that you could call, enter a PIN, and dial the number you want to reach.

3. International Callback

Finally, an international callback solution acts as another strong alternative to VoIP in countries with VoIP bans.

Businesses in restricted countries would dial a trigger number, hang up the call, and then receive a callback. Upon receiving the callback, they would enter the destination number to call and connect.

How United World Telecom Can Help

At United World Telecom, we provide VoIP solutions and alternatives to VoIP that work well in blocked countries. To learn more, contact us and chat with our experts online!

A system outage or any type of downtime can significantly impact your company’s ability to close sales or deliver customer support. This may result in a loss of business continuity and revenue. So, maintaining a cloud phone system with a high VoIP uptime and network redundancy is essential to your business’ ability to function.

Are VoIP Systems Reliable? Why VoIP Uptime is Important

Since a redundant architecture is critical to business operations, how reliable are cloud-based communications?

You may be surprised to hear that VoIP networks are more consistent than the wired setups of the PSTN. Landlines are susceptible to power outages, severe weather disruptions, and phone line damage – causing lengthy downtimes. This is because landlines have a single point of failure, meaning your entire system stops working when it fails.

IP telephony eliminates wired setups. Instead, all you need to get started is a stable internet connection. And even if your internet goes out, you can simply forward calls to another destination. So, cloud-based systems continue working during power outages, inclement weather, and more – making them more reliable than traditional phones.

While hosted solutions are certainly more trustworthy than landlines, it’s still critical to consider a provider’s VoIP uptime. Without a high uptime guarantee from your provider, you’re at risk of serious downtime. And downtime means tough consequences such as losing customers, profits, productivity, and more.

What Exactly is VoIP Uptime?

In other words, it’s a metric used to determine the provider’s quality of service, performance, and reliability. In contrast, “downtime” indicates the time a service spends offline, out of operation, or “down.”

You can usually find uptimes listed on your provider’s website, in their Service Level Agreement (SLA), or by contacting them directly. To determine VoIP uptime, most providers either use a monitoring service or measure it themselves using simple math.

It’s essential to confirm that your provider includes both scheduled (maintenance, upgrades, etc.) and unplanned (disasters, power outages, etc.) downtimes in their assessments. This way, you know their uptime is accurate.

What Does 99.99% Mean?

Many cloud providers use percentages to communicate their VoIP uptime to customers. These percentages typically start at 99% and go up from there. However, providers can’t guarantee true 100% uptime. This is because networks are susceptible to damage from natural disasters, cyberattacks, and other unforeseen technical issues that cause service interruptions.

Now, you’re probably thinking – If I can’t have 100% uptime, 99% is exceptional. However, while 99% is undoubtedly better than 75% uptime, it can still yield substantial downtime for your business.

In fact, every “9” in an uptime guarantee is significant. It lets you know how much downtime to expect during a given year. For example, 99% uptime is equivalent to 3.65 days of downtime per year. Whereas 99.99% (also known as “four nines”) is only 52.60 minutes of annual downtime.

Here’s a table listing the range of VoIP uptime percentages and how much downtime you should expect with each:

How Do Providers Ensure 99.99% VoIP Uptime?

It’s important to understand how providers ensure reliability and guarantee 99.99% availability. This way, you know exactly what to look for when choosing a provider.

• Global Points of Presence (POPs)

Having several global points of presence allows providers to establish a high availability service. If one server fails, providers can redirect traffic or fall back on a different POP. So, you never experience downtime if your provider experiences an outage.

For example, if your provider’s Seattle-based POP failed, they could instantly route call traffic from the Seattle server location to another US-based POP.

• VoIP Security

While VoIP is fairly secure as is, your provider might face security risks like Denial of Service (DOS) that compromise their system and lead to downtime. So, it’s necessary to look for a trustworthy provider that meets the minimum security requirements for your specific communication and industry needs.

Search for providers with voice encryption security like Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP) and those with regular system security checks and firewalls. Providers with these security practices are more likely to deliver a consistent and secure business phone system with a high VoIP uptime.

• Networks of International Carriers

Selecting a provider that partners with multiple international carriers yields similar benefits, as discussed in the POP section. Working with numerous global carriers allows providers to reroute calls seamlessly to alternate destinations if one carrier fails. So, your provider (and your business) never experience prolonged outages or downtime. This helps providers build a more reliable and redundant service with high availability.

What Uptime Guarantee is Right For Your Business?

The answer to this question depends entirely on your business operations, communication setup, and service demands. 99.999% uptime is the industry gold standard – as it only allows about 5 mins of downtime per year.

But when determining the right uptime for your business, it’s helpful to ask yourself:

• How integral is your cloud phone system to your critical, everyday operations?
• How much downtime can your business afford?

Then, based on your responses, select a provider with your preferred VoIP uptime guarantee.

Best Practices for Developing a System with High VoIP Uptime

While high availability falls mainly on your provider, it’s still up to you to build a redundant system and minimize downtime for your business and customers. Let’s review the best practices for developing a system with high uptime.

1. Carefully select your provider

Selecting a trustworthy provider with the right VoIP uptime for your business is crucial to developing a consistent network. When researching cloud telephony providers, evaluate the following factors:

• • Availability guarantees
  • Redundancy methods (multiple POPs and carriers)
  • VoIP security measures (voice encryption, firewalls, regular security checks)
  • Disaster recovery and business continuity plans.

2. Double-check Service Level Agreements (SLA)

Many providers list their uptimes on their websites, which may not always be accurate. So, it’s important to double-check a provider’s service level agreement. This document will list their true uptime percentage and permitted downtimes. The SLA may also list terms and conditions for what happens if their actual uptime dips below their guarantee.

3. Monitor your system

Regularly monitoring your VoIP system allows you to track any network issues causing interruptions. Simply start by integrating an open-source or paid monitoring service into your existing setup. Then, you’ll be able to quickly identify and troubleshoot your VoIP system.

4. Implement failover strategies

With failover strategies in place, you’ll build your network’s redundancy. Even if your business experiences outages or crises, your system will continue to work with failover methods. Consider the following:

• • Use failover forwarding to route calls to alternate locations if the first destination is unavailable
  • Set up a backup or alternate internet connection
  • Opt for hardware that is simple to repair or includes a warranty
  • Disperse your teams, servers, and other systems across multiple geographical locations
  • Backup all mission-critical data, documents, applications, and so on
  • Create an in-depth disaster recovery plan and use it to train your teams.

Get High VoIP Uptime and Consistent Service with United World Telecom

Through a highly redundant global network, United World Telecom offers 99.999% VoIP uptime to users. We deliver high-quality telephony solutions with high availability as we have built long-term relationships with multiple Tier-1 global carriers. Additionally, we have several international POPs and decades of experience in telecommunications.

To learn more about our redundancy measures and if we are the right provider for you, speak with our dedicated telecom experts. Call us at +1 (561) 908-6171 or chat with us online today!

Good call quality is essential when communicating with customers and prospects. This is because it’s a first (and rather important) impression of your business and its reliability. 40% of customers stated they stopped doing business with a company due to poor customer service, and this includes a bad caller experience.

Part of customer service is being able to communicate on the phone without any hiccups or interruptions like muffled audio or dropped calls. So, how can you achieve the best VoIP call quality for your business? Let’s take a look!

Improving Call Quality for Your VoIP Phone System

VoIP providers exist in abundance these days. Some are fly-by-night using least-cost routing (LCR) with low-quality termination. Others offer expensive plans with limited global coverage and no redundancy.

Along with low-quality cloud phone service providers, many factors affect good VoIP call quality. Part of improving voice quality over your phone system is correctly identifying the issues and resolving them.

Common VoIP Issues

In order to improve VoIP call quality, you need to be aware of the different issues affecting voice quality. These are the most common voice over IP issues that you’ll need to troubleshoot:

• Inability to make calls from a device
• Dropped calls
• No sound
• Jitter
• Echo or delays
• Broken or muffled audio
• Latency
• Packet loss

It is crucial to identify what VoIP problem your system faces. It helps you take the right course of action to improve your network’s functionality. Finding a high-quality VoIP phone service provider can take care of most of these issues. That aside, there are a few tweaks you can make to your system on your own. Here is a VoIP troubleshooting guide to get you started.

How to Achieve Best VoIP Call Quality?

Take these steps to achieve good VoIP call quality for your cloud phone system:

1. Measure your VoIP call quality
2. Ensure you have the right equipment
3. Troubleshoot VoIP quality issues
4. Disable SIP ALG
5. Troubleshooting with VoIP QoS
6. Switch to a new provider

1. Measure Your VoIP Call Quality

It is vital to constantly monitor and measure VoIP call quality. Mean Opinion Score (MOS) measures VoIP quality by scoring a call from 1 (bad VoIP quality) to 5 (excellent VoIP quality). A good VoIP call quality MOS score falls between 3.5-4.2.

Here’s an example of what a MOS score chart looks like:

Test and measure call quality by monitoring jitter, latency, and packet loss. Additionally, call detail records and activity reports will help you identify where your call quality might be lacking. Then, you can determine how to resolve the issue.

2. Ensure You Have the Right Equipment

Purchase the right equipment to support your call volume and agent requirements. Here is a list of equipment needed for VoIP:

• VoIP (hard phones or softphones)
• Computers or laptops
• VoIP phone service
• Headsets with a microphone and noise-canceling technology
• High-speed and reliable data that meets essential VoIP bandwidth requirements

If you are a United World Telecom customer, you can work closely with our telecom experts to understand the equipment needed for your specific phone system. Chat with us online!

3. Troubleshoot VoIP Quality Issues

Besides finding the necessary equipment, it is crucial to test your VoIP phone system periodically. Here are some ways to troubleshoot VoIP call quality issues on your own:

1. Check your internet speed
2. Review network and equipment configuration
3. Identify and resolve jitter, latency, and packet loss issues
4. Check cables connecting devices and replace faulty wiring
5. Invest in good headsets
6. Avoid hubs on your local access network (LAN)

If you still suffer from low or bad call quality after trying these troubleshooting methods, then the problem may lie in your VoIP service.

4. Disable SIP ALG

Most internet routers come with the SIP ALG feature activated. SIP ALG — also known as Application Layer Gateway — prevents problems caused by a router’s firewall. It does so by monitoring and modifying VoIP traffic, as needed, to improve connectivity.

However, the downside is that VoIP traffic (voice data packets) can get lost due to router firewalls when transferred between the phone and the VoIP provider. This can lead to:

• One-way audio
• Phone not ringing for incoming calls
• Calls sent to voicemail directly
• Dropped calls

So, if your VoIP issues persist after trying these troubleshooting methods, you may need to check and disable SIP ALG.

Disabling SIP ALG depends on the type of modem you use. For most routers:

• 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 router settings are not clear, reach out to your internet provider for help.

5. Troubleshooting with VoIP QoS

VoIP Quality of Service (QoS) is the process of addressing and fixing voice quality issues. QoS prioritizes network traffic passing through a router. The end goal is to offer users acceptable-to-good service.

Routers handle data (and voice) packets on a ‘first in, first out’ basis. This way, all traffic on every device within your network is considered a priority. When too many people use the internet simultaneously, these packets end up waiting in long queues. This means, your call quality competes with other internet traffic on your network — leading to low voice quality.

QoS balances out bandwidth distribution across different types of traffic and devices. This way, you can prioritize what traffic gets through sooner and hassle-free to achieve the best VoIP call quality.

Set up VoIP QoS by reaching out to your internet supplier, VoIP providers, and IT team. To learn more, check out our guide to VoIP QoS standards and requirements.

6. Switch to a New Provider

Finally, if the above methods don’t work, it might be time to find a new provider to meet your business’ exact needs. Sometimes, no matter how you try to improve your call quality, the problem may lie in the phone system itself. Research different cloud phone service providers to understand how their system and phone network works. Consult with telecom experts to find the right solution for your business communication needs.

Resolving VoIP Call Quality Issues

Achieve the best VoIP call quality for your business by trying these different troubleshooting methods. And if those don’t do the trick, reach out to your VoIP provider.

Speak with our telecom experts to understand how we can help you achieve good VoIP call quality for your business. Call us at 1 (877) 898 8646 or chat with us online!

Has your business considered upgrading its legacy ISDN phone system to gain new features and capabilities? Then, you must consider an ISDN to SIP migration.

SIP trunking is cost-effective and highly scalable. And this makes it an excellent choice when upgrading your phone system.

Here is a brief guide to replacing ISDN with SIP trunking, including the benefits of an ISDN to SIP migration and the different steps involved to make this transition smooth and cost-effective.

Why Switch to SIP?

SIP phone systems are often considered a modern alternative to traditional business phone systems. By replacing ISDN with SIP systems, you can add more flexibility and scalability to your phone system.

It is a good practice to upgrade legacy phone systems to take advantage of updated and advanced technologies. Here we will look at how you can migrate to SIP and what an ISDN to SIP plan looks like.

How Can SIP Trunking Benefit Your Business?

There are many benefits of switching to SIP trunking, from making your phone system cloud-friendly to aiding in global expansion to bringing more flexibility.

SIP trunking uses session initiation protocol (SIP) to send voice, video, text, and other unified data through the internet instead of traditional POTS lines. With this ability, you can run your business communications from any location and connect with your team wherever they are located.

And the best part? You don’t need additional hardware or new equipment to get your SIP system running. Global SIP trunks can easily integrate within your existing PBX. So, you do not need to install a new system. Simply bring SIP trunking into your current system. Additionally, your business can even expand to new markets and regions without increasing overhead costs.

So, with SIP trunking, you get a more advanced and scalable phone system while keeping your communication-related costs low.

What to Keep in Mind When Migrating to SIP Trunking

When doing an ISDN to SIP migration, note that:

• You will need to conduct an audit of your current phone system to gauge the scope of the migration.
• Since SIP can work alongside ISDN, switching won’t cause any downtime or interruptions.
• Tests that ensure the system runs smoothly can be conducted around your business’ schedule and activity.
• Research different SIP trunking providers to get a sense of what they offer and what their rates are.

Debunking SIP Trunking Myths

When it comes to new (and complex) technology, it is easy to get sucked up in misinformation and doubts. And so, before we discuss how to migrate from ISDN to SIP, let’s go over some common SIP trunking myths and discuss them:

• SIP call quality is poor — When it comes to digital communication, there is a tendency to believe that communicating over the internet can cause packet loss, latency, and network jitter. But SIP trunking uses dedicated IP connections designed and optimized for high call quality. Because of this, they offer the same, if not better, call quality than ISDN calls.
• SIP trunking comes with hidden costs — SIP trunking is a cost-effective alternative with two primary costs: monthly fees and call costs. You choose the plan at the beginning of your contract. With the plan, you can make and receive local and international calls for low costs. And there are no installation or set-up fees involved. In fact, with some providers, you won’t even pay a cancellation fee, if you choose to find another provider. One additional fee might be to upgrade your PBX system and internet bandwidth if you have low internet capacity. But even with these costs, you will find the service ROI-friendly.
• SIP trunks are not secure — Another common misconception is that cloud communications (like other internet services) come with cybersecurity risks and can be easily hacked into. However, with advanced technology, cloud communication, and SIP trunking providers offer a great deal of security. Additionally, you will need to work with your internet service provider to ensure firewalls and security measures are in place.
• Migrating to SIP will result in downtime — As we mentioned before, many are afraid that switching services or communication systems can lead to interruption and downtime. However, with SIP trunking and a quality service provider, you can minimize VoIP downtime and migrate without pausing your service.

ISDN to SIP Migration in 7 Steps

So, what does an ISDN to SIP plan entail and how can you prepare for such a migration? Here is a 5-step plan to replace ISDN with SIP trunking:

1. Identify and Document Your Requirements

Before you get started, you need to identify what you need from this migration:

• Do you need new business phone numbers? If so, how many numbers do you need?
• What types of numbers do you need: local or toll-free?
• Do you need call management features such as cloud IVR, auto-attendant, call queues, call monitoring, and so on?
• What locations do you need calls forwarded to?
• Do you need extensions or dedicated numbers (also known as direct inward dialing numbers) for specific employees or departments?
• Do you need a softphone to support remote workers?

It is also a good practice to document what you currently have and want to keep, such as phone numbers, extensions, voicemail inbox, etc.

Once you determine what you need from your new system, you can start planning for the SIP migration.

2. Communicate with Your Employees

Before making the switch, bring your employees into the migration process. Get their input and feedback to identify additional areas of conflict and improvement. Some questions to ask employees include:

• What do you think of our current business phone system?
• Have you encountered any issues using this phone system? If so, provide details.
• What would you like from a new phone system? Explain why.
• Can you think of anything that will improve internal and external communication?

3. Make a Plan and Scope the Job

The next step to replacing ISDN with SIP is to make a plan. This will help you determine what your business should expect in terms of scope, transition, and cost. Depending on the state of your current system, you may have to transition in parts (floor by floor or branch by branch) or migrate the entire system in one go.

Work with your communications or SIP trunking provider to get an idea of what the transition will look like, and how your business can prepare. When preparing your plan, consider the following:

• Employees — List down all staff members and collect their contact information.
• Vendors and suppliers — List down contact information for essential vendors, suppliers, and service providers.
• List all communication equipment — Create an inventory of all business-related phones, laptops, computers, desk phones, fax machines, smartphones, etc.
• Call routing — Identify what numbers will route calls to which numbers and locations and when.
• Backup data — Make sure information and data are backed up and that backup processes are working properly.
• Setup safekeeping for critical data — Keep critical data extra safe as you may need this to enact disaster recovery and business continuity. Think logins and passwords, alternative numbers and phone and lines, etc.
• Include remote working capabilities — Here’s an opportunity to explore remote working options for your employees. Review how your new system can support this flexibility.
• Create and test a business continuity plan — This plan will help your business continue working and offer services if something goes wrong.

4. Conduct an Audit: Evaluate Existing Systems

Next, evaluate your current system and anticipate possible problems and changes required. Look at the infrastructure and bandwidth capacity. What is your current system lacking that a SIP phone system can solve?

Check with your SIP provider how much bandwidth and network capacity you will need for your business’ call volume. This information will help you decide if additional connections are required to prioritize voice traffic over other internet usages. Similarly, communicate your call volume expectations with your provider. They will help you understand what infrastructure will work best for your business needs.

Lastly, you may even want to let your provider on future expansion plans. This will help you determine if this provider’s service is scalable and can assist your business in growing when it is ready.

5. Get and Install a SIP Trunking Service

Once you’ve completed the audit, the next step is to find a reliable SIP trunking provider and subscribe to the service. Research different providers to get a sense of what they offer and how much they charge.

You should also look for additional features and tools they offer that can further enhance your business communication system; for example, call management features like call routing or softphones, and so on. Another important factor in choosing a SIP trunking provider is their support team. Is their support team easily accessible? Do they offer 24/7 customer support, in case your business needs help off-hours?

Once you’ve settled on a SIP provider, it’s time to set up and get your SIP system up and running. One significant benefit of replacing ISDN with SIP is that SIP trunking requires less equipment and hardware. And since most SIP services are hosted and maintained by your SIP provider, you have little to no installation. You might need to tweak and adjust minor things, and you might choose to install the service section by section. This ensures no downtime or interruption.

Finally, prepare for the ISDN to SIP migration by getting the right equipment ready.

6. Test Your New SIP System

As you set up your SIP phone system, you must test it and its different components to ensure everything is working smoothly. Most SIP systems can integrate with other business applications such as CRMs, video conferencing tools, APIs, and so on. Test your system with core business applications to identify any issues or hiccups. Similarly, your provider should test the service and phone numbers before the system goes live. Keep your business continuity plan ready, in case there are any issues and you need a backup.

7. Go Live and Make Business Calls

Finally, once your SIP system has been tested and any issues have been resolved, you can fully switch over to SIP. The installation and activation of the system is relatively quick; it is the preparation that can take some time. However, with the right SIP provider and the right tools, this transition will occur smoothly, and your business communication system will work effortlessly.

Ready to Switch Over to SIP?

If you are looking for ways to upgrade your existing business phone system without increasing expenses, then SIP trunking is the way to go. Want to learn more? Our experts at United World Telecom can help you understand how SIP trunking works and how it can help your business specifically. Call us today at 1 (877) 898 8646 or chat with us online!