8 Pre-Launch Steps To Go With The WebRTC Based
Application Development
Table of Contents 1.
Introduction
2.
Before you begin
3.
The factors
4.
3.1.
Compatibility with device
3.2.
Compatibility of the browser
3.3.
Issues regarding device permission
3.4.
Network problems
3.5.
Recording a session
3.6.
Unable to switch
3.7.
Increased latency
3.8.
RTMP live streaming
Wrapping it up
Introduction Numerous companies have been using internet-based communication mediums for some time. With the outbreak of the COVID-19 pandemic, the user base of video calling and chatting applications increased even more. Naturally, it isn’t surprising to see entrepreneurs and the owners of small businesses invest in the development of WebRTC systems. There’s much to gain from such a solution for a business owner, after all. Experiencing and interacting with people via live video has become quite popular in numerous applications across various industries. With real-time engagement through live video, hundreds and thousands of businesses managed to transform the way they operate. Today, business enterprises, telemedicine platforms, online education platforms, entertainment & sports platforms, virtual event platforms, e-commerce, and everything else in between, use the power of WebRTC technology to establish communications in real-time via mobile applications. 3
If you’re searching for a WebRTC app development company, then it’s assumable that you have a project you wish to delegate to the professionals of a digital agency. Now, it won’t be wise to start development without making a few crucial decisions. These decisions will dictate the abilities your application will have, as well as the user experience you hope to bestow upon your clients. Furthermore, your decisions will ensure a profitable future for your deployment and the effort you need to put into maintaining and updating the services you’ll provide through your application. That’s why this topic will include information about the points that require your attention when you plan your WebRTC project. You’ll also learn about a couple of common issues associated with this technology and how to solve them for improved communicability in real-time.
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While you’re going to employ the services of a WebRTC app development company, it’s assumed that you’ll build your solution with an open-source WebRTC system. Then again, you should take a moment to explore PaaS platforms, such as AWS Chime and Agora because they can offer much more scalability, as well as advanced RTC SDKs. They are compatible with a massive variety of browsers and devices. As a result, the factors highlighted in this write-up will point out how PaaS platforms like Agora manage live videos.
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Before You Begin Now, before you hire the best providers of web real-time communication services, you need to look into scalability as it has a vital role to play in making a WebRTC project successful. The Mesh architecture allows every participant in one video room to send and receive audio, video, and data concerning their connection. They can share these things with all the other participants. It means if you connect to five different individuals, you’ll have five open connections. The problem with WebRTC is that it won’t scale properly if you use it this way.
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If you allow more users, you’ll have to deal with bandwidth and CPU problems. There’s a solution to this issue, of course. You’ll need a media server that can allocate the video among participants. By using this strategy, you’ll get the opportunity t use more advanced features, such as integrating your WebRTC solution with other types of technologies. You’ll also be able to process the video streams. Media servers will help you resolve the participant scalability issue, but the power of these servers has limitations too. At some point in time, you may have to resort to another server, apart from the one you already have. If the number of participants increases, you’ll need to maintain the media server, which can be a challenging task. If media servers don’t do the trick, you must turn to API platforms.
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API platforms are nothing but a collection of servers and development kits for clients. They can give you everything you require for building a WebRTC solution. All the API platforms will come together to perform the basic functions on the server-side, including session connections, signaling the parties, and maintaining the flow of media across different network address translations and topographies. These APIs have multi-party recording, communications, streaming, as well as third-party integrations to support identity management and other abilities. Conversely, the API platforms on the client-side SDK will provide support to normal mobile devices and browsers designed specifically for desktops. Fundamentally, these platforms allow you to focus on the business logic only. The API platforms will be in charge of the complex tasks of scaling and maintaining the WebRTC solution. They give you the client-side SDK, apart from setting up the media server to ensure you can execute your WebRTC project effectively. 8
Agencies offering web real time communication services suggest using the most common API platforms for WebRTC that can support and scale almost one million partakers. Just remember that your business use-case will define this feature. ●
Twilio
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Agora
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Red5Pro
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Vonage
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Here are a few other widely used open-source API platforms. ●
Janus
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Pion
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FreeSwitch
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Jitsi
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MediaSoup
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Asterisk
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Kurento
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The Factor WebRTC development specialists will remove most of the complexities associated with building such an application. Nevertheless, you’ll have several decisions to make while handling multiple moving parts. At the moment, it won’t be a mistake to say that this WebRTC technology has the power to uplift just about any business. It’s one of the main reasons why innumerable businesses spread across the world are already using it, including budding startups and already-established MNCs. However, everything boils down to the way you implement this system practically. If you want to do it the right way, you should pay attention to the following factors.
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1. Compatibility with device In the sphere of mobile devices, WebRTC solutions aren’t as popular as in the field of desktops. As mobile gadgets work uniquely, top rated webRTC developers need to create something different. Just consider a hypothetical situation for a minute. One individual is using an iOS phone for video chatting and another individual joins it using Google Chrome. In such circumstances, communication problems will appear sooner rather than later. It’s possible to fix the issue if you can pinpoint the device at the earliest stages and move forward accordingly.
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2. Compatibility of the browser All browsers don’t have WebRTC features. At least, they can initiate those features simultaneously. Similarly, a few browsers may allow WebRTC features to function in one but not in another. In such a situation, you can choose to display a warning message to your users to check whether WebRTC functionalities will be accessible in that specific browser or not. In doing so, you’ll manage the situation appropriately.
3. Issues regarding device permission When designing and developing a WebRTC solution for the world of the web, you can use APIs to gain access to smartphones and computers having microphones and cameras. You may end up with device permission problems, especially if one of the participants doesn’t permit another device to communicate with in real-time. You can solve this matter on the pre-call screen. 13
4. Network problems Inept networks affect WebRTC solutions profoundly. Even the work of the top rated webRTC developers will go down the drain if the network connection isn’t stable enough. Users will inevitably encounter problems where both video and audio calls take place perfectly in one place but degrade swiftly in another. You can resolve this issue if you optimize the bandwidth and set up an appropriate codec for the video. With Agora SDK, you can build a few functions that will facilitate bandwidth optimization and deliver a smooth experience.
5. Recording a session You won’t be able to provide a recording service if you use the WebRTC API for browsers. When it comes to recording a session, you’ll need to set up a media server first. After all, the server-side will be in charge of managing the recording requirements. With Agora SDK, you can offer a collection of REST APIs that will allow users to record and save the session to use later
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6. Unable to switch There will be situations where you have to render various layouts based on the way a business uses WebRTC. If you resort to various UI components for unique layouts, you may not be able to give your users a seamless experience. You should use only one UI component for all layouts. In doing so, you’ll bolster the user experience.
7. Increased latency If you’re going to use the core API of WebRTC for your project, you’ll have to put up with quite a bit of latency during streaming. Latency depends on the processing capabilities of the media server, as well as its efficiency. For providing the best of everything to your users, you should consider using third-party systems with previously optimized latencies for streaming. With Agora, extremely low latencies are a guarantee. 15
8. RTMP live streaming It will be a challenge to append a URL for live stream into a video conference using WebRTC. For instance, if you hope to append your live video released on Vimeo into a WebRTC conference, you’ll experience problems. Fortunately, with Agora SDK, you’ll have an option to include this functionality.
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Wrapping it Up When it comes to building WebRTC solutions, the platforms mentioned in this write-up, including Agora, should be part of your tech stack. You already know about the features Agora has to offer. So, consider hiring an app development agency that utilizes Agora to build WebRTC solutions.
Original Source: https://www.moontechnolabs.com/blog/pre-launch-steps-with-webrtc-based-application-development/
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