The products and services in this review roundup are focused on business use and because of this either provide some PBX features or serve as full-on virtual PBXes. This may mean, among other things, that they provide service to telephone sets on your employees' desks. Most also support electronic faxing in some fashion, either directly (which can be a significant challenge for some VoIP services) or by simply integrating an incoming fax with your email system. Other popular features are video conferencing and shared meeting software (so meeting attendees not only hear each other but can present presentations or documents in a shared work space).
Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries without a central database, like the UK, it might be necessary to query the GSM network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of least cost routing options, it needs to provide a certain level of reliability when handling calls.
Overall, VoIP is simply the better option for the vast majority of customers. Dropping your landline means no more hidden fees or metered long distance calling charges. Everything is charged at one low rate by most providers and your ability to customize your phone service to exactly what you need is far greater. Unless you've got some highly unique circumstances that somehow mandate a landline, VoIP is simply the better choice.
Similar to Ooma's residential service (below), AXvoice deploys its home VoIP with the help of an appliance, appropriately called the AXvoice Device, which sits between your home's phones and your Internet router. This device not only serves as a bridge between your old phones and the new VoIP service it also enables many of the advanced features that straight POTS bridges often don't address.
The RTCP extended report VoIP metrics block specified by RFC 3611 is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal/noise/echo level, mean opinion scores (MOS) and R factors and configuration information related to the jitter buffer. VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP summary report or one of the other signaling protocol extensions. VoIP metrics reports are intended to support real-time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
If you are seeking a superb pan/tilt/zoom (PTZ) video conferencing camera for medium-size meeting rooms, the Yealink VCC20 is an ideal choice. 2.18M pixels HD CMOS sensor plus 12X optical zoom power the VCC20 HD PTZ, delivering HD images up to 1080p at 30 fps. The Yealink VCC20 camera is an optimal solution for a smaller business environments. It is so detailed and true to life that you feel like you are in the meeting room with everyone else.
Typically, price is one of the most important reasons people opt for residential VoIP. One of the most attractive is the "triple play" sales pitch we mentioned above made by almost every regional residential cable company and internet provider: Get your Internet, TV, and phone service all rolled into one monthly charge. Not only is that usually an attractive number, it also means a technician will hook everything up for you including your phone, and you'll probably be able to use the same phone you're using now instead of having to migrate to a VoIP phone.

It's also critical that you consider the impact of mergers and acquisitions on your phone system, both from your own organization's perspective as well as your VoIP provider. Because VoIP systems turn calls into data, the whole process isn't as plug-and-play standards-based as the old-fashioned analog phone system might have been. Should your company merge with or purchase another, VoIP compatibility will become another significant IT issue.
VoIP can be used for free with computers and even, in some cases, with mobile and landline phones. However, when it is used to completely replace the PSTN service, then it has a price. But this price is way cheaper than standard phone calls. This becomes thrilling when you consider international calls. Some people have had their communication costs on international calls cut down by 90% thanks to VoIP.
In the following time span of about two decades, various forms of packet telephony were developed and industry interest groups formed to support the new technologies. Following the termination of the ARPANET project, and expansion of the Internet for commercial traffic, IP telephony was tested and deemed infeasible for commercial use until the introduction of VocalChat in the early 1990s and then in Feb 1995 the official release of Internet Phone (or iPhone for short) commercial software by VocalTec , based on the Audio Transceiver patent by Lior Haramaty and Alon Cohen, and followed by other VoIP infrastructure components such as telephony gateways and switching servers. Soon after it became an established area of interest in commercial labs of the major IT concerns. By the late 1990s, the first softswitches became available, and new protocols, such as H.323, MGCP and the Session Initiation Protocol (SIP) gained widespread attention. In the early 2000s, the proliferation of high-bandwidth always-on Internet connections to residential dwellings and businesses, spawned an industry of Internet telephony service providers (ITSPs). The development of open-source telephony software, such as Asterisk PBX, fueled widespread interest and entrepreneurship in voice-over-IP services, applying new Internet technology paradigms, such as cloud services to telephony.
GoToConnect has nearly as many glowing critical reviews as the Radiohead discography, minus the pretentious lyricism. GoToConnect has established a positive industry reputation since its launch as Jive in 2006, thanks mostly to its interface simplicity, focus on small businesses, and large array of telephone features that are available to all pricing tiers.
also enables T56A  to work as a base station which can be registered with up to 4 compatible Yealink handsets. This solution provides you with a quick and reliable DECT connection without wiring or cabling. As a complement for Yealink DECT series, attaching DD10K to your desk phone offers you a new solution by combining the desk phone’s features with DECT capabilities.

In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside India.[61] This effectively means that people who have PCs can use them to make a VoIP call to any number, but if the remote side is a normal phone, the gateway that converts the VoIP call to a POTS call is not permitted by law to be inside India. Foreign based VoIP server services are illegal to use in India.[61]
Packed with advanced phone system features necessary to operate a small to medium businesses, such as hosted PBX capabilities, desk-to-desk calling, automated attendant systems, call routing and even music-on-hold, Business phone services powered by VoIP technology make it easy for any company to operate with the same level of professionalism customers expect from large scale enterprise systems. Business VoIP systems also include overall Unified Communication solutions to empower the mobility and flexibility needed for any size businesses. With an inexpensive, feature filled phone solutions, your business can operate at a high level on par with large scale systems, without having to worry about the cost.
Using a separate virtual circuit identifier (VCI) for audio over IP has the potential to reduce latency on shared connections. ATM's potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM VCs. The latest generations of DSL, VDSL and VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic.[16]
That being said, Grasshopper doesn’t offer any conferencing tools. For that, you’ll have to sign up for join.me—Grasshopper’s sister company. This service offers both video and audioconferencing, but it does cost an extra $10–$30 per month. That’s another strike against Grasshopper, since most providers in Grasshopper’s price range include conferencing features.
Businesses working remotely should keep in mind, though, that Ooma doesn’t offer any videoconferencing tools. Instead, you get a conference line that lets at-home employees collaborate over the phone. That puts Ooma a pace behind some of the other providers on our list, but with providers like Nextiva currently offering their video collaboration tool to businesses for free, it shouldn’t be a major issue.

The products and services in this review roundup are focused on business use and because of this either provide some PBX features or serve as full-on virtual PBXes. This may mean, among other things, that they provide service to telephone sets on your employees' desks. Most also support electronic faxing in some fashion, either directly (which can be a significant challenge for some VoIP services) or by simply integrating an incoming fax with your email system. Other popular features are video conferencing and shared meeting software (so meeting attendees not only hear each other but can present presentations or documents in a shared work space).
Your company needs real time access to manage your phone system in or out of the office. Our online interface makes it possible to manage your system from anywhere with voicemails, call logs, call recordings, and call routing being just a click away. If you don't have time to make technical changes, our support staff are available for all your needs.
In addition to making sure your internet service can handle your VoIP traffic, you also need to make sure your local area network (LAN) can handle it. What makes network management tricky with VoIP is that if you simply drop it onto your network, that traffic will get processed the same as any other traffic, meaning your shared accounting application or those 20 gigabytes worth of files your assistant just stored in the cloud.
Businesses working remotely should keep in mind, though, that Ooma doesn’t offer any videoconferencing tools. Instead, you get a conference line that lets at-home employees collaborate over the phone. That puts Ooma a pace behind some of the other providers on our list, but with providers like Nextiva currently offering their video collaboration tool to businesses for free, it shouldn’t be a major issue.

PhonePower is one of a handful of VoIP providers that actually specialize in residential VoIP rather than business VoIP. Although PhonePower has many plans, it’s best for calling within the US (including Puerto Rico) and Canada. That’s because it has possibly the cheapest prices in residential VoIP, providing you’re calling solely on local numbers. PhonePower also enables calls abroad, although there are cheaper options such as Vonage if you’re planning on making more than an hour’s worth of calls internationally each month.


Our editors have researched and tested hundreds of systems, filtering out industry leading business phone services with the highest levels of reliability, backed by unparalleled customer service, and aggressive price points. The small business VoIP providers we've featured below offer custom packages for any budget, dedicated support reps, competitive pricing, and a fully managed, hands-on approach to getting your new business VoIP system up and running in the shortest possible time. Compare these providers below, some of which are from our partners, to find the right one for you.
In 1999, a discrete cosine transform (DCT) audio data compression algorithm called the modified discrete cosine transform (MDCT) was adopted for the Siren codec, used in the G.722.1 wideband audio coding standard.[74][75] The same year, the MDCT was adapted into the LD-MDCT speech coding algorithm, used for the AAC-LD format and intended for significantly improved audio quality in VoIP applications.[76] MDCT has since been widely used in VoIP applications, such as the G.729.1 wideband codec introduced in 2006,[77] Apple's Facetime (using AAC-LD) introduced in 2010,[78] the CELT codec introduced in 2011,[79] the Opus codec introduced in 2012,[80] and WhatsApp's voice calling feature introduced in 2015.[81]
Back-end integration with custom and third-party apps, like CRM systems, also open a whole new world for your calling data because now it can extend the phone system beyond just basic voice communication. Such integrations allows users to transfer calls to and from their mobile phone, place and receive calls from their personal phone (that appear to be coming from the business), and interact with colleagues and customers via voice and text -- all from a variety of devices. But it also allows recording and analysis of call data to measure things like customer satisfaction, understand your sales audience at a new level, and even handle customer requests and problems automatically without the customer ever being aware they never spoke to a human.
1. The Microsoft 365 Business Voice service components of Domestic Calling Plan and Audio Conferencing are sold inclusive of all required taxes and fees, including 911 fees and other transactional taxes that typically apply to communication services in the U.S. The price includes these taxes and fees until June 30th, 2021. The Phone System component is sold tax exclusive and any applicable sales tax will appear as a separate charge in the U.S.
Similar to its popular small business VoIP solution, Ooma Office, the company touts its on-premises VoIP appliance to power its residential service. You'll find three versions of this device to choose from: the Ooma Telo, Ooma Telo Air or Ooma Telo 4G, but they all sit between your Internet router and your phones, making installation of this low-cost service plug-and-play.  
Communication on the IP network is perceived as less reliable in contrast to the circuit-switched public telephone network because it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental Quality of Service (QoS) guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to data loss in the presence of congestion[a] than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.[16] Therefore, VoIP implementations may face problems with latency, packet loss, and jitter.[16][17]

However, VoIP traffic to and from the on-premises systems can often also be sent over secure private links. Examples include personal VPN, site-to-site VPN, private networks such as MPLS and SD-WAN, or via private SBCs (Session Border Controllers). While exceptions and private peering options do exist, it is generally uncommon for those private connectivity methods to be provided by Hosted or Cloud VoIP providers.


While understanding the basics of VoIP and SIP is important, setting one of these systems up will require some general network knowledge, too. For the best quality, you will need to meet a minimum upstream and downstream data throughput requirement. In addition, you'll also need to meet a minimum latency number (that is, the time between when a signal leaves a remote computer and when your system receives it), typically measured in milliseconds. It is possible to test your network connection to see if it will support a VoIP service. RingCentral offers this service from their website, other vendors like to have their service engineers do it for you.  
Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new Private branch exchange (PBX) lines installed internationally were VoIP.[9] For example, in the United States, the Social Security Administration is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.[10][11]
The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Packet delay variation results from changes in queuing delay along a given network path due to competition from other users for the same transmission links. VoIP receivers accommodate this variation by storing incoming packets briefly in a playout buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout, i.e. momentary audio interruptions.

Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new Private branch exchange (PBX) lines installed internationally were VoIP.[9] For example, in the United States, the Social Security Administration is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.[10][11]
SIP is built to work on a peer-to-peer (meaning endpoint to endpoint) basis. Those two points are called the "user-agent client" and the "user-agent server." Remember that those points can be swapped, so that in SIP, the endpoint making the call is the user-agent client initiating the traffic and endpoint receiving the call is the user-agent server receiving the call.
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