WebRTC And FreeSWITCH – What This Combination Means?

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.

Browsers awaiting their turn

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.

WebRTC is on the way

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.

The working process

So, how do customize webrtc application solutions work? Here’s a breakdown of the process.

• The preferred browser of a user will connect to one of the numerous web servers available to load a website containing JavaScript on all pages.

• The JavaScript on the page will take charge of controlling the media interfaces of the browser, including cameras, microphones, speakers, and everything else. As a result, it creates a media object API.

• The media object of the API of WebRTC will have all the abilities of all gadgets and available codecs, including the sample rate, etc. In turn, it’ll allow the user to select capabilities and preferences.

• The pages of a website will interface with the user of the browser while getting some kind of input to sign in to the communication services of the webserver.

• The JavaScript will utilize the signaling method it can lay its hands on, whether XMPP, custom, SIP, or proprietary over an encrypted and secure WebSocket. It will do so to sign in to the communication service, find peers, and originate or receive calls.

• Once a user signs up to the service, he/she will be able to make or receive calls through the customize webrtc application solutions. The signaling system will give the address of the protocol of the peer.

• Now, it’s time to ascertain the real IP addresses. The JavaScript will create a WebRTC object API to pinpoint its own IP addresses, ports, and transports. The peer will receive it to be able to exchange media.

• At this point, the net API of WebRTC will exchange the ICE candidates with one of the users until both of them find the appropriate IP address triplets, transport, and port for every stream.

• As soon as they acquire the perfect addresses, the signaling will set up the call.

• Once the communication signaling with the peer is ready to perform, media capabilities start getting exchanged in SDP format. The two peers, of course, have to agree on the media formats.

• When they agree upon the media formats, JavaScript WebRTC API Transport will utilize secure or encrypted WebSockets as the mode of transportation for data and media.

• The WebRTC JavaScript API media will kick into action to render the streams of media received by the recipient.

• Additionally, or as an alternative to the current media format, peers can set up one or several data channels. Through these channels, they’ll exchange structured or raw information bi-directionally.

• Once they hang up, signaling will break the call and JavaScript Media WebRTC API will kick into action to shut all renderings and streams down.

 

Also Read: 8 Pre-Launch Steps To Go With The WebRTC Based Application Development

 

Security or encryption

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.

More than peer-to-peer communication

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.

Gateways and servers

If you’re going to hire developers, you need to make sure they can solve the only problem with WebRTC – implementing a signaling plane of some kind. It’s also about implementing an all-inclusive SIP stack of signaling in JavaScript. Then again, in terms of the network, as well as the media plane, WebRTC is the only compatible system with the existing world of telecommunications. It utilizes similar concepts and techniques.

FreeSWITCH is a combo of all

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.

Final words

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.