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A VoIP phone is a kind of telephone that uses IP technology to transmit calls. It can come either in the form of specialized digital hardware or a program (running on a computer or mobile device) that performs the same functions.
With its microphone and receiver, a VoIP telephone takes the sound you generate and converts it into packets of data, which it sends over the network and out through the internet. On the other end, the phone decompresses the data and plays it back for the other person to hear.
In the case of business phone systems, a traditional phone is connected via a wire to an on-premise private branch exchange (PBX)—the bulky equipment that allows the system to connect all internal extensions. In this office phone setup, the PBX is essentially the brains of the system since it manages all routing and ensures all calls reach their destination through the public switched telephone network (PSTN).
PSTN simply refers to old-school telephony, in which phones are hooked up to specific networks—primarily used for landlines. You may have also heard of plain old telephone service (POTS). The two acronyms refer to the same thing, with the latter being a colloquial term for the former.
With an IP phone system, you no longer have to keep VoIP and IP PBX hardware inside your office premises. You can choose to have it hosted so that you’ll get all the enterprise-level functionality you need in a phone system without having to worry about the overhead costs of maintaining a server inside your building.
That’s because modern VoIP-based cloud phone systems ditch the limitations of traditional telephony, allowing computers and other connected devices to place phone calls through the internet. It is able to do so thanks to transport protocols, which are responsible for establishing the connection and making sure data packets reach their destination.
Before you can understand how VoIP telephones transmit calls over the internet, you have to first understand how VoIP technology works.
Voice over IP technology is actually a set of different protocols working together to replicate telephony functions. Each protocol has a different function but all are working simultaneously in real-time.
The general purpose of SIP is to set up real-time multimedia sessions between two or more participants over the internet. In VoIP, this protocol is the signaling component of the technology. It:
While there are more complex processes going on to establish these connections, that is basically what SIP does for VoIP.
RTP, on the other hand, is the one responsible for carrying multimedia; which in the case of VoIP is voice audio, over the internet. It carries the digital voice audio data packet back and forth during VoIP phone calls. After which, codecs then convert the compressed digital audio data packet to uncompressed audio so that it can be played or heard by the call participants. There are many different types of codecs, which can affect call quality and clarity. RingCentral, in particular, supports G.722 audio codec and OPUS audio codec for HD voice.
Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) are the two primary types of transport protocols used for data transmission across the internet. How does one differ from the other?
As previously mentioned, data travels over the internet in packets. These data packets are transmitted using the IP network. Imagine sending a letter—you need an envelope to make sure it is delivered to its destination untouched and without delay. TCP and UDP are two kinds of envelope you can use, and both carry the voice signals in the form of data packets.
TCP is a connection-based protocol module that offers some good integrity- and error-checking capabilities, which is why it’s commonly used in most web-based traffic. Between the two protocols, TCP is more reliable because it’s designed to cope with network failures and can adapt to available resources in the network.
When a packet is sent from one end to the other, the destination acknowledges receipt by sending a packet back to the source. If the source doesn’t receive an acknowledgment packet or if the packet states that there was a problem during transmission, the packet will be resent.
Sometimes, it takes a few seconds for a packet to be fully transmitted. This accurate delivery of information is the reason TCP is typically used for websites and emails. However, problems come when the packets get lost or distorted. This causes time delays, and user experience can take a hit.
TCP is so prevalent across the internet that it’s usually combined with Internet Protocol (IP) and written as TCP/IP (TCP over IP).
While TCP is more optimized for accuracy, UDP is focused on speed. It is connectionless so the data packets can be sent without any “negotiation.” In a sense, it’s almost like a “send and forget” protocol, as it provides just enough control information to understand what app is running and to check whether the packets got distorted in transit.
The process ensures swift transmission. But since there’s little error checking involved, it’s likely for packets to get distorted along the way. For VoIP calls, this could mean a slight slip in words, choppy voice, or distorted audio.
Exceptional VoIP technology does not require a completely reliable transport protocol since errors like packet loss or distortion only have a minor impact on voice signals. You should be better off dropping a packet and have a few milliseconds of silence than to have seconds of lag.
To illustrate, imagine being in a call—you hear something a second or two after the person on the other line actually utters the words. Contrast this with simply missing a syllable of sound but being able to fill in the gaps through context clues—it’s arguably less frustrating.
While there are many VoIP service providers that employ only one type of protocol, truly good VoIP companies like RingCentral use both to ensure the best call quality and efficiency possible.
In network terms, a port is a communication endpoint used by the transport protocol to facilitate the stream of information. Every single piece of data has a destination port associated with it, and this port is associated with a number to enable the server to sort traffic and deliver the data to the correct recipient.
Imagine a seaport where numerous cargo ships dock on a regular basis. Each ship is assigned a berth number corresponding to its designated spot in the seaport to avoid confusion. Think of the berth number as the network port number, which provides a fast and easy way for the network to identify its recipient.
Port numbers range from 0 to 65535. For VoIP traffic, the standard port is 5060, which is used for both TCP and UDP connections. Consider it the gateway for VoIP functions such as media streaming and video conferencing. Often, port 5004 UDP is preferred when transmitting packets of data within a computer network.
In the case of VoIP providers, they use a variety of ports based on their specific product designs and requirements. They may also have a supplementary list of UDP ports for when they need to upgrade their products.
Apart from network ports, you may also need hardware ports in order to use complementary VoIP accessories like a headset for devices that are not designed to be used like regular telephones. If you have to use a USB phone, you need a USB port.
While these are not all the components of VoIP technology, these are the main protocols and elements that work with each other to allow for voice calls over the internet.
VoIP phones are designed to provide users a regular telephone-like experience by implementing these protocols in real-time. This allows IP phones to deliver IP telephony to users.
The viability of VoIP for businesses rely on its ability to replace traditional telephone carriers. If VoIP-based solutions are only able to establish calls with other VoIP-based telephones, then it will not work for most organizations.
While residential VoIP has caught on and is rapidly being adopted by regular consumers, most people are still using traditional carriers for both landlines and cell phones.
These are the same people that businesses are targeting, so if VoIP cannot connect to regular landlines and cell phones, then it is a no-go for many companies.
Fortunately, VoIP can connect to regular landlines through gateways.
VoIP gateways allow digital voice calls from VoIP phones to be converted to analog or multiplexed voice data that can be consumed through regular landlines and cell phones. In turn, it is also able to convert analog multiplexed voice data into a digital format so that it can be broken into IP packets, which can then be transmitted through VoIP.
As businesses all over the world continue to embrace cloud communications, you should find ways to enhance your internet phone experience. You can make the most out of VoIP technology with any of the hardware listed below.
Better yet, turn to a reputable cloud communications provider like RingCentral. It and companies like it will expertly assess your needs so you save on capital and operational costs. Hosted VoIP solutions call for very little hardware, which means fewer expenses on your end. Because the provider manages your hardware (and software), you can relax knowing you are safe from equipment issues and unimaginable maintenance costs.
A router is responsible for forwarding data packets between devices in a network or from the devices to the internet. To ensure the data packets don’t get lost within the network, the router designates a local IP address to every device.
If your company spends a lot of time communicating with customers and suppliers, you are extra vulnerable to information leakage and cyber attacks. In this case, a router is your network’s first line of defense.
Businesses take security into account when setting up their network infrastructure. The larger ones usually have designated IT professionals to make sure it remains secure. But if you sign up for a hosted VoIP service, everything is handled by the provider.
Consumer-level routers should be able to support about a dozen devices. But if you need something that could handle more computers and provide increased security and additional traffic for voice and data streams, it is a must to invest in a business-class router. After all, growth should be in every business owner’s mind.
Originally, “handset” was the term used to describe the device you pick up and hold close to your ear and mouth so you can listen and talk with the person on the other line. In a regular landline phone, you will see the handset connected by wire to the unit.
Along with the advances in business communications technology, “handset” has also been used to refer to the part of the phone—whether wired or wireless—that delivers the same function. In the case of mobile phones, the entire unit can be referred to as the handset.
A VoIP handset typically has the same basic features and capabilities as that of a traditional phone. After all, it delivers the same function, but with help from the internet. Sure, you can use VoIP technology in many other ways (instant messaging, web conferencing, video conferencing, etc.). But perhaps depending on the situation, nothing compares to the experience of lifting a handset to place a call.
With this VoIP adapter, you can place calls over the internet even if you’re using a traditional landline device (and be able to use the same number).
Yes, it’s possible! ATA acts as an interface between your analog phone and your VoIP system, converting analog signals into digital traffic. At first look, this seems like a huge money-saving practice, especially if you want to keep your existing PSTN phones and don’t have enough budget to purchase new VoIP phones.
But the thing is, ATA does not give you access to all the advanced features built in to modern VoIP handsets. As such, some businesses use ATA only as a transitional device until they are ready to purchase their own IP-enabled phones.
By installing an app (more on this later), you can use your computer or mobile device as a VoIP phone. You get the full functionality of a “regular phone” without having to purchase one. This is a common practice among businesses, as it offers mobility and usability unseen in traditional phone systems. It also gives your employees flexibility because they don’t have to be tied to their office desks all the time.
Apart from the wrong prioritization of QoS, dropped calls can also be attributed to insufficient bandwidth. This is especially the case if dropped calls are frequent during the busiest times in your office.
There are different kinds of VoIP phones available, and each has a specific purpose. Sure, a desk phone would work in a conference setting—but you could be missing important features without a conference phone. This is something to consider if you plan to have a full VoIP deployment for your company.
Softphones are basically software apps installed onto your computer or mobile device so they can be used like a hard phone. In essence, they are virtual phones that are not tied to any physical location. You can connect a USB phone to your computer so you can leverage the softphone’s functionality and features and communicate as you would using a regular phone. Often, softphones running on a computer and coupled with a headset deliver better call quality than VoIP apps on smartphones. But generally, aside from the computer or mobile device, there is no need for other devices or physical phones that require a power adapter or an ac adapter.
A VoIP phone is the digital transport vehicle for phone calls and is, therefore, an integral part of your organization’s network infrastructure. Imagine still using a traditional phone system when almost everyone else—including your competitors—has switched to VoIP. You don’t want to get left behind. Consult RingCentral and learn what kind of VoIP phone would work well with their VoIP-based unified communications solution while meeting the needs of your business!