VoIP, SIP and webRTC, what make them different and why it is important?

4/13/2022 - 6 min. read

A voice call is being made

Let’s imagine those situations: You have big news you have to share with your loved ones on the other side of the country, or your credit card is blocked in the middle of a shopping spree or you want to get customer support because your new computer turned into a toaster. What is the same thing you want to do in those cases? Talk to a person, even if they are not physically next to you, preferably now!

So you grab your phone, dial a number and get access to the person you want to talk to, instantly. That’s the power of the voice. So far, that’s the closest thing we have to instant teleportation and that is why voice is great and has been around for more than 100 years now. Since it started a lot of technology shifts happened. The magic of a call remains the same but the means have changed and this article is to clarify certain things regarding today’s technologies.

The State of Voice

Indeed it comes to voice technology (or “voice” for short), you can hear a lot of different things and be a bit confused about the terms used and what they refer to. Since we have been working in the voice industry for quite a while at Cantoo let us clarify what RTC, VOIP, SIP, and WebRTC terms refer to and what technology does what.

Before jumping into the details, let me just go back in time a bit and remind you (or let you know) that in the 2000s we saw the beginning of an important switch in terms of voice services. You all remember the good old phone you had at home that was plugged into the wall and that was cutting your internet connection when ringing. This was RTC voice technology. It has been around forever, it is the first massively widespread telephony technology and it was doing fine until internet and IP communication arrived. Indeed, you see, RTC didn’t really have a contender before and it was quite costly to produce and maintain for telecom operators. Also and as a consequence, it was quite expensive for the end-user, on top of not being very flexible in terms of service and setup. So when the new VOIP, or voice over IP, kid arrived in the neighbourhood, it changed the status quo.

Thank you internet for the SIP/RTP

With the development of the internet and the spread of IP networks, a cheaper more efficient, and easier way of transporting voice appeared: carrying calls over the IP network. To do so basically two things were necessary, signalling protocol and media carrying protocol. For those who are not familiar with these concepts, to establish a call one operator has to signal to the other, its intention to send a call and ask for permission to do so from the receiving operator. Once it has received the green light from this receiving party to send a call the media protocol establishes the voice flow transmission (this is when the ringing stops and you start talking). When the conversation is done and someone hangs up, the signalling part takes over again and the two operators terminate the call by sending goodbye signals to each other (telecoms are well mannered). The two protocols at play here are SIP for Signaling and RTP for the media flow.

So when you hear that someone produces a SIP service or creates a SIP trunk, they are producing a voice over IP service and this service uses the SIP protocol. We make this precision because there is another way to produce voice services: webRTC. Although the name refers to the good old RTC it is another Voice over IP solution.

App and web-ready: the Web RTC

SIP/RTP voice technology is in many ways more efficient and easier to produce than RTC technology but it remains a telecom technology which means it requires a bit of work and knowledge to use it. You need to set up a proper telecom infrastructure to deal with the flows and with the signalling, you need to understand the basics of the telecom technical interconnections, you need to manage technical infrastructure components, and their interaction with each other, and so on.

So it is still very telecom and it remains quite complicated for a web actor who is just willing to add a call capability in its app for instance, because it requires the said actor to learn a whole new technology. So to bridge the gap between telecom and web technology, webRTC comes in handy. Indeed WebRTC uses HTTP as a protocol so it is way easier to incorporate it via an API in one’s app or website. It only requires a few lines of code to be able to generate a webRTC call and it’s great when you want to start using telecom means to improve your user experience. Nevertheless, with the easiness of webRTC comes drawbacks. HTTP is not a protocol that has been designed for voice and although it works fine, it is not optimized for it. Therefore, quality, reliability, and performance can become an issue when using webRTC at scale. As one grows its telecom usage, one has to look for a more stable and optimized way of routing its increasing amount of calls, and this is exactly what SIP/RTP technology was built for.

In addition to the previously explained webRTC limitations, WebRTC is not the way operators and carriers interconnect with each other in the voice business. You can’t have 100% webRTC communication between two people who are on different telecom networks because at one point you will have to go through the public telecom network and webRTC is not designed for such interconnections. It is nevertheless possible to have 100% webRTC communication between two people as long as they are on the same network. For instance, two people calling through Whatsapp may have a 100% webRTC communication since it stays within the Whatsapp network. As soon as you have to reach another network you have to talk in the common tongue of the telecoms, and this is massively SIP/RTP.

The best of the two worlds combined

So what happens when a call is generated in webRTC and has to be terminated to the public telecom network? We convert the call, it is transcoded from webRTC to SIP/RTP and then send to the receiving operator with the SIP/RTC protocol. It requires a bit of processing power to do so, but it’s usually ok. At Cantoo since we enable telecom services with webRTC for our clients and we talk in SIP/RTP with our operator partners we transcode the calls and allow our clients to reach the world by combining the easiness of webRTC with the power of SIP/RTP.

In summary, SIP/RTP and webRTC are Voice over IP (VOIP) technology. WebRTC is the easiest way of integrating voice services in a product, it’s handy, quick, and easy, but it needs its big, powerful, and high-quality brother SIP/RTP to be able to talk to other networks and terminate calls around the world. Those two technologies are complementary and usually serve a different purposes and a different level of calls volume. You start with webRTC and start to enjoy the power of telecoms and move to full SIP/RTP technology when your volume of calls increases. When investing time, people, and money into new technology is justified by your traffic volume.

Whatever your volume of calls and the technology you want to use, at Cantoo we are happy to support and help you improve your voice services, without financial barriers, technical constraints, and in total transparency.

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