As a business owner using a Virtual Phone System, you are reaping the benefits of the professional quality service of a business system combined with the ease and cost-effectiveness of a single phone in your hand. But you may face some challenges using this system well.
Voice over Internet Protocol (VoIP) systems were never intended for home or small business service, after all, and consequently, users may face hiccups when using these systems that way. Fortunately, recent changes to VoIP are addressing many of those problems.
For the first time, Talkroute is offering a VoIP-based service with WebRTC because we believe the technology is finally stable enough to offer a reliable internet-based call connection. Here’s why we’re using WebRTC to go beyond old-school VoIP and why that can make your business communications faster, stronger, and simpler.
The Challenges of VoIP
VoIP does many things well. Without it, you’d be forking out a lot of money for a mammoth phone bill every month. That’s what businesses and families had to do until about 15+ years ago.
Basically here’s how VoIP is solving the long-distance problem for you..
Remember when your parents limited long-distance phone calls to a few minutes because they were so expensive? That’s because your voice had to travel over hundreds or thousands of miles of copper wires using a switched telephone network. As long as you and whoever you were talking to had those particular wires tied up with your conversation, no one else could use them. You were renting those wires in essence, and the phone companies were charging you a pretty penny to do it. In economic terms, you were demanding, and Ma Bell was supplying. And she was the one making money.
Then came VoIP. VoIP happened due to the telecommunications act of 1996 that mandated that smaller carriers be allowed to compete with the RBOCs. So VoIP introduced competition to telecommunications, and Ma Bell, who had kept American phone systems in a vice grip, suddenly had to compete with third party carriers. Thanks to VoIP, calls that were expensive because there was no competition, were getting cheaper.
Plus, VoIP introduced new technology. Instead of all those convolutions of wires, VoIP uses a piece of hardware called a driver and a type of software called codecs (coder-decoders) to transform your voice into binary data. Then, it breaks down that data into even tinier bits called packets,and transfers those packets from the computer through the router over the wires and into another computer where the process gets reversed. It may sound complicated, but actually, this transfer happens much faster and more cleanly than the old analog system could do it. (More on this later.)
Do You Have to Sacrifice Voice Quality?
But VoIP is not a perfect system.
For one thing, VoIP doesn’t always feature great voice quality. If you have a strong internet connection, it’s probably fine. But if your connection is weak, slow, or unstable, you may get jittery or broken audio. Some users solve this problem by buying “jitter buffers” that temporarily store arriving packets and thus can minimize delay variations.
Delay or latency also crops up with voice. Delay happens when speech takes an unusual amount of time to go from the speaker’s mouth to the receiver’s ear. Usually, it sounds like an echo. Delays occur for different reasons. One, it takes electrons a full second to travel through 125,000 miles of fiber. If you are talking to someone halfway around the world, that means a 70 millisecond delay. Packets can also get delayed when a device doing the handling either malfunctions or gets overloaded.
Security is another concern with VoIP. Everything with VoIP goes through the internet, so if your business is staying on top of its cybersecurity, you’re probably fine. But if you get breached, your voice calls — as well as your written materials — are fair game for the cyber criminals. You may have to encrypt or secure your internet before using a VoIP system in order to protect your business and your customers’ data.
Finally, for VoIP to work, your internet has to be up and running. Without internet access, you don’t have a phone system. Now, that’s probably not an issue for most businesses most of the time, but in some rural areas with spotty coverage, it can become a concern.
WebRTC can help solve many of these challenges and improve on the good things VoIP has already brought to modern telecommunications.
What Is WebRTC?
WebRTC (Real-Time Communications) is an open source project that enables web browsers and mobile devices to engage in real time communication. Essentially, it’s a collection of APIs (application programming interfaces) that permits browsers to communicate with each other directly. The browsers can use WebRTC to exchange any type of data. All the major web browsers such as Google, Mozilla, and Opera support the WebRTC project’s framework.
WebRTC is a suite of protocols and tools that uses websockets to allow real-time communication applications to be created easily, and it enhances websocket technology. Websockets alone have the ability to communicate peer to peer or server to peer. WebRTC is not required for that.
How Does WebRTC Work?
Everything goes through the API. From there, the server identifies the client and in this way helps keep the connection secure. At that point, the server needs to determine the type of data it’s sending — video, voice chat, text, or something else. Next comes NAT (Network Address Translation) traversal, which is a technique that establishes the connection between the two clients.
NAT is an IP protocol. Its basic function is to transfer metadata from the browser. One of its most common uses is in your home router. NAT allows multiple computers to share one internet connection in your home or office. Actually, one of the biggest barriers for VoIP is NAT because SIP and RTP alone weren’t designed to work with it.
WebRTC, however, has NAT traversal tools built into it, which greatly improves the reliability of calls. WebRTC also employs automatic encryption to maintain security, so if someone listens in to the transfer of data between the clients, they won’t be able to obtain any useful information. Finally, the system determines how the data will get compressed and transferred.
What Challenges Does WebRTC Face?
WebRTC uses UDP (user datagram protocol), which isn’t 100% reliable when transferring critical data. It’s fast and consistent, but it doesn’t follow up on itself. Basically, UDP never makes sure the client receives the data, just that the data is sent. However, this can actually be a benefit when you’re doing something like video chat. Lose a few frames? No one will even notice.
Another challenge is that WebRTC doesn’t have standard signaling protocol. This means that different developers have different methods of implementing protocols. And lastly, it’s still not compatible with all browsers. WebRTC, however, is compatible with Google and most of the other major ones, including Mozilla, Chrome, Firefox, and Opera. No plug-ins needed. Safari and Edge, however, do require those plug-ins.
Like any new product or service, WebRTC still needs some work. But at Talkroute, we believe the technology has reached the stage of its evolution where it will begin to offer our customers much better service than traditional VoIP.
Why Was WebRTC Developed?
Without WebRTC, things like live-streaming are just too unreliable. WebRTC also removes the need for extra apps like Skype or Zoom. You can do the same things you used to do with these apps directly from the web browser thanks to WebRTC. Another benefit is that the server doesn’t need to use any more resources to process incoming requests since clients can talk peer to peer.
Talkroute is now using WebRTC technology to bring its powerful voice platform directly to you without needing to rely on a phone line.
But Isn’t WebRTC VoIP?
Short answer? Yes. Long answer? It’s more complicated than that.
Voice calls over WebRTC are not VoIP as most of us traditionally know it. Instead, WebRTC enables users to place and receive VoIP calls in a far more reliable manner than traditional VoIP does. Even on low-quality internet connections, WebRTC uses a technology called Websockets to reliably send and receive calls over the internet.
What’s the Problem with Traditional VoIP and How Does WebRTC Improve on It?
SIP and RTP
Traditional VoIP usually uses two key technologies. One is called SIP, and the other, RTP. SIP, which you’ve most likely heard of, is a signalling protocol. By way of a simple explanation, when you place a call from your traditional VoIP phone, it sends a message using SIP to a server. In a SIP message, there are a few key pieces of information: Where the call is coming from, who the call is going to, and how to reach me (the phone).
RTP is the second part of the equation and is less well known than SIP. RTP is what’s responsible for carrying your voice from your phone to the server that’s processing your call. RTP listens to the SIP messages and sends your voice to wherever SIP says it is supposed to go. This configuration faces several challenges, which WebRTC improves upon.
When SIP and RTP messages go out, they’re blindly sent out to the internet without first securing the connection. This can cause these messages either to be incorrect or not to reach the server at all. WebRTC establishes a dedicated connection on the internet first by using a piece of technology called a “Websocket.” After the Websocket is established, SIP and RTP messages can be sent without risk of error.
Network Address Translation
Traditional VoIP was never designed to be used with home and office internet routers. Internet routers (specifically a protocol called NAT [Network Address Translation]) introduce a major challenge in order to allow traditional VoIP to work correctly over them. Common problems routers introduce are:
1. Calls with one-way or no audio at all
2. Your phone not ringing when a call comes in
Since WebRTC establishes a Websocket first before any call signalling occurs, these issues are completely mitigated.
To further complicate things is something called a codec. To simply explain, a codec is how your voice is translated into a digital signal so it can be sent over the internet. Traditional VoIP uses fairly old codecs, usually PCM (think .wav file). PCM is a very heavy codec and is easily susceptible to interference that occurs on the internet.
Talkroute uses a new and revolutionary codec called Opus that allows VoIP calls to occur with a fraction of the bandwidth. The Opus codec not only allows calls with a tiny amount of bandwidth, but it also has the potential to allow calls of even higher quality than traditional VoIP. Opus even has the ability to send equal and sometimes higher quality audio while using a fraction of the bandwidth traditional codecs require. What does that mean for you as the user? It means that Talkroute’s implementation of WebRTC can work well even on very low quality internet connections.
If you’d like to dig more deeply into this subject, G.711 on this chart shows PCM/ULAW, traditional VoIP quality
WebRTC vs the PSTN
PSTN is an acronym for Public Switched Telephone Network. If you place or receive a call using a phone number, that call is taking place on the PSTN. Talkroute has been and always will be a service that pushes the PSTN to its limit. When Talkroute forwards a call to your phone, it is using the PSTN. However, with Talkroute’s desktop app, you can place and receive calls using WebRTC, which does not rely on the PSTN.
Is WebRTC a Silver Bullet?
No. However, WebRTC is a dramatic improvement in reliability over traditional VoIP. Talkroute’s WebRTC utilizes Websockets and Opus to enable these improvements. It is still possible to experience call quality and reliability issues with WebRTC, but many of the issues traditional VoIP experiences are mitigated by WebRTC. Want to know more about how Talkroute’s new WebRTC service can help your business advance? Contact us.
We hope you found this article insightful and were able to learn more about the technology powering our new desktop app!
The desktop app is available for Mac, Windows, and Linux operating systems and can be downloaded here.
Stephanie is the Marketing Director at Talkroute and has been featured in Forbes, Inc, and Entrepreneur as a leading authority on business and telecommunications.
Stephanie is also the chief editor and contributing author for the Talkroute blog helping more than 100k entrepreneurs to start, run, and grow their businesses.