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VoIP

Learn about voice over IP (VoIP) services and how businesses can use VoIP technology to communicate better with local and international customers.
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14th January 2021 Call Management, Learn, Voip

PRI Explained: What is a Primary Rate Interface?

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Understanding Voice Over IP Jitter, Latency, and Packet Loss

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PRI Explained: What is a Primary Rate Interface?

Posted on by Beny Mouen

Source: DepositPhotos.com - Lic#12814344ID#27446420

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

Posted on by Luke Genoyer

Source: DepositPhotos.com - Lic#1497111 ID#27446420

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

Posted on by Miguel Isabal

Source: DepositPhotos.com - Lic#216688154 ID#27446420

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

Posted on by Miguel Isabal

Source: DepositPhotos.com - Lic#103222742 ID#27446420

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.

webrtc softphone

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

Posted on by Beny Mouen

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?

Posted on by Nadine Sampson

Source: DepositPhotos.com - Lic#199689480 ID#27446420

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?

Posted on by Beny Mouen

Source: DepositPhotos.com - Lic#182873428 ID#27446420

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

Posted on by Nadine Sampson

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!

Why VoIP is a Good Fit for Distributed Teams

Posted on by Miguel Isabal

Source: DepositPhotos.com - Lic#44591887 ID#27446420

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!