Overview: Implementing WebRTC: 6 Key Pointers
Web Real-Time Communication (WebRTC) is the go-to technology for live streaming applications and WebRTC Video API has made peer-to-peer streaming easy and convenient. WebRTC being a powerful technology allows developers to transfer data swiftly in real time with low latency. Because of which, it has been considered as the best technology to build video chat apps among developers.
So, if you are a developer who wants to build a video chat app and are a WebRTC enthusiast, then this article is for you!
What is WebRTC API?
WebRTC API is a technology that allows developers to set up peer-to-peer connections. It can be integrated in any type of application and enable the APP to capture media, encode and decode audio and video, transport layers and manage sessions.
How does WebRTC Video API function?
For the WebRTC to function, Session Traversal Utilities for NAT (STUN) and Traversal Using Relays around NAT (TURN) servers are required.
WebRTC functions on a request sent by the public-facing IP address to a STUN server on which the remote server responds with the IP address it sees. It is like a computer making a query to a remote server, questioning what is the IP address it receives the query from.
WebRTC functions to enable APPs and sites capture and stream audio and/or video content , as well as to exchange arbitrary data between browsers without requiring an intermediary
Using a WebRTC API developers get a ready-made solution that doesn’t need any plugins or third-party software.There are plenty of linked APIs, protocols, and signaling-related technologies that work in sync because of WebRTC API.
3 steps that Developers need to follow to enable WebRTC communication to function on their respective applications:
1. getUserMedia() is the gateway into a set of APIs that provides the means to access the user’s local camera, microphone and various other media streams.
2. RTCPeerConnection JavaScript API must be able to share basic network information like the ports, IP addresses, etc via signaling.
3.RTCPeerConnection JavaScript API must also disclose basic information about media data.
Industry-Based Examples of WebRTC Video API:
The rising popularity of WebRTC stack is because the technology has made modern communication simple and effective. Major audio and video applications such as Google Hangouts, Facebook Messenger and many others use WebRTC API as it is easy to develop, is secure, flexible as well as scalable.
Here are some industry-based examples of WebRTC:
E-commerce
Real-time videos support retailers to communicate with possible customers and display products in a more organized and appealing manner.
Virtual Consultations
Mainly used by consultation professionals such as doctors, using in-app live video chat apps offers the luxury of getting answers from anywhere and at any time.
Virtual Recruitment
After the pandemic, virtual recruiting has become the best way-to-go! Thanks to the virtual recruiting environment and tech-driven interviewing techniques offered by APPs.
Healthcare
Telehealth has become the next big thing in the healthcare industry! Several apps now offer comfort to patients so that they can reach out to their doctors as well as expand the doctors’ access to appointments for patients that are not able to make it to the clinic for any specific reasons.
Insurance
Video-enabled insurance claim procedures can improve claimant’s satisfaction and adjusters’ productivity.
Here Are The 6 Factors You Need to Analyze Before Implementing WebRTC:
Undoubtedly, WebRTC opens up a huge window of opportunity. However, with any kind of technology, the final choice needs to be analyzed and requires careful consideration.
Such as whether to choose an MCU or SFU topology, server-side or client-side recording, etc. Here are six factors you as a developer should analyze before taking the first step towards implementing WebRTC.
1. SFU or MCU?
WebRTC itself can only provide peer-to-peer communication in the browser. Conferencing support requires an intermediate server to receive and transmit media data. This goal can be achieved with two topologies: Multipoint Conferencing Unit (MCU) and Selective Forwarding Unit (SFU). The right choice between MCU and SFU topology is very important to ensure communication quality.
2.Record from server side or client side?
Recording is not an essential part of the WebRTC infrastructure. Enabling this feature requires careful consideration, such as whether you should use server-side or client-side recorders.
3.Connect VoIP or PSTN?
VoIP may be very effective as a managed network as it offers best capacity control, elastic extension, and quality calls, it may be a problem on an IT network that is not managed by local issues such as home and bad weather. Therefore, if it is invited, you must dial the Active VoiP based WebRTC session in the PSTN.
4.Measure The Risks
WebRTC is used by a trusted security architecture. However, the ecosystem of WebRTC consists of many customers, hosts, servers, applications, and transportation layers, and cyber security threats. Hence, it is always good to measure the risks.
5.Selecting the right codec
Choosing the right video codec is important because WebRTC projects allow live streaming without installing plugins. Therefore, a thorough analysis of the WebRTC video codec is worth careful consideration.
6.Understanding WebRTC Stack Limits
It is true that RTC was not designed for group calls and has always been peer-to-peer oriented from the start. This is why scalability has always been an issue with the “vanilla” WebRTC protocol. So, before joining WebRTC, it’s important to understand the limitations to avoid future disappointments.
Considering all these complexities, it is wise to outsource your WebRTC implementation to a vendor rather than opting for a DIY approach.
Do All CPaaS Providers Offer the Same WebRTC Solutions?
If you decide to outsource, do all CPaaS providers offer the same technologies when using the WebRTC protocol?
It is important to note that not all CPaaS solutions are similar to each other, even though they are based on similar technologies. Many technologies are used on top of the WebRTC protocol, which is a big difference.
Other CPaaS solutions can differ significantly in many respects, such as bandwidth adaptability that depends on simultaneous transport, single or multiple RTCPeerConnections, in addition to the impact of existing users who will continue to use PSTN/VoIP, etc.
Resources to Develop Video Calling APP with WebRTC:
Here are the four resources to consider while developing a video calling app with WebRTC.
- Develop Multiparity Video chat Application for iOS
If looking for building a video chat app for iOS, use iOS Toolkit and WebRTC platform Video APIs for calls. This has an advantage of ramping up development by hosting on their devices.
- Build Multiparity RTC on PHP
To build a multiparity video chat application on PHP, use 3rd Party JavaScript Toolkit and an app server of your choice.
- Develop Video Call on Flutter
Flutter APIs are loaded with powerful conferencing, collaborative and reporting features. Create a fully functional 1-1 or group video calling with Flutter using Video Embed.
- Build Video Call on Android
To build one to one or multiparity real-time video calling apps on Android, use Android Toolkit and platform server APIs.
Conclusion:
WebRTC is an open source technology that has changed the way real-time communication happens. It has made data sharing and peer-to-peer conferencing easier than ever before, without users having to install any plug-ins or third-party software. WebRTC consists of several interrelated APIs and protocols which work together to achieve this.
