Completed 12 Years
Last Updated: 06 Oct 2021 | 9 min read
WebRTC has been around for a long time, and you probably know a thing or two about it already. If you have been enjoying the advantages offered by WebRTC to your business, you’ll probably appreciate it if another exceptional system gets integrated into it and augments it even further. FreeSWITCH has got that honor.
WebRTC has been around for a long time, and you probably know a thing or two about it already. Maybe you even use it in your business. However, WebRTC also happens to be one of those tech fields that never got any enhancements. FreeSWITCH came up and fixed the situation.
FreeSWITCH finally added something new to WebRTC. This fact is actually quite refreshing, especially for those who work at a webrtc app development company. After all, they now have the chance to learn and experiment with it again. This change took its sweet time to arrive too – at least a decade. However, as soon as it came, it allowed WebRTC to take that quantum leap. In fact, the combination of WebRTC and FreeSWITCH disrupted the entire sector of communication.
There’s already a massive base of installers of web browsers on both computers and smartphones consisting of hundreds and thousands of people. Sooner or later, the number will hit billions. These browsers are nothing less than full-fledged terminals of communication. They support both video and audio endpoints without extra software programs, hardware elements, plug-ins, or anything else. Today’s web browsers have everything an individual needs to interact with microphones, loudspeakers, screens, headsets, cameras, and every other device by default.
A web browser is nothing but the new endpoint of the present day and age. It’s also powerful enough to represent a phone, a CPE, etc. Web browsers have APIs and they get updated automatically. They’re even compatible with just about every system that currently exists under the sun. No one needs to configure, procure, upgrade, or support a browser. They’re always ready to help you reach out to new clients with new services, and they’re suitable for businesses of all types, sizes, and scales.
When it comes to communication, it has two entirely separated flows. These include media and signaling. The former is the digitized content sent and received via communication. It can be anything from video files, audio files, screen-sharing, and more. On the other hand, the latter is the information flow that defines the person calling the other person, as well as the paths they take. It also represents the technology used in transmitting the content types mentioned above.
These two flows usually follow unrelated paths while going from the caller to the recipient. For instance, the IP packets travel through different routers and gateways. Additionally, separate software programs manage the signaling and media using unique protocols.
A digital product designed by a webrtc app development company defines the way a browser gains access to its intrinsic media-capturing ability. It also defines the way it delivers and receives files from peers through a specific network, as well as the way it renders the stream of media received. It does so by using Session Description Protocol or SDP, and the method of operation is just like SIP.
In other words, WebRTC includes media content of all types. It never prescribes any signaling systems, either. It’s entirely a design-related decision embossed in the basic definition. Some of the most recognized signaling systems are XMPP, SIP, and custom or proprietary protocols. WebRTC is also about encryption. It encrypts all media streams as a mandatory method of operation.
Firefox, Chrome, and Opera are the most widely used browsers right now. They occupy 70 percent of the market share of web browsers, and they’re already implementing the standards of WebRTC. Microsoft Edge also announced that it’s going to incorporate the primary features of WebRTC. Apple’s Safari, however, isn’t budging at all.
So, how do customize webrtc application solutions work? Here’s a breakdown of the process.
As the user, you need to note that in regular operation, everything will get encrypted via real PKI certificates from genuine Certification Authorities, real DNS names, HTTPS, SSL, DTLS-SRTP, TLS. It’s precisely the way it should work. Security isn’t an afterthought in WebRTC. It’s a necessity.
WebRTC is nothing but a technique used by browsers to deliver and receive media through the internet, peer-to-peer, or maybe with the support of a relay server, only if they can’t reach each other directly.
After going through the information provided above, you’re probably wondering where FreeSWITCH fits into WebRTC. Well, FreeSWITCH is an enhanced version of WebRTC. In fact, it implements all low-level protocols of WebRTC, other than the necessary requirements and codecs. It has everything, including SRTP, RTP, DTLS encryption, WebSocket, and secure WebSocket transportation. As it has everything, it has the power to cater to SIP endpoints through WebRTC, and it relies on mod_sofia to do so. Through another mod, called mod-jingle, it interacts with XMPP.
Basically, it’s better for you to hire developers specializing in FreeSWITCH if you want the best of all worlds. The creators of FreeSWITCH designed it specifically to manage, as well as message high-def media, including audio and video. Several years ago, FreeSWITCH started receiving support for the OPUS audio codec as a pioneering feature. It has also been evolving continuously over the years to be exceptionally robust and self-healing to be able to sustain losses equivalent to more than 40% packets. It even maintains understandability. In simple terms, FreeSWITCH can be pivotal in your WebRTC project.
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