WebRTC Technology for Video Streaming App Development: Use Cases and Best Practices

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In this article, we discuss how WebRTC technology is used in apps. Find out why WebRTC is one of the most popular choices for developing video streaming purposes.

What is WebRTC?

WebRTC (Web Real-Time Communication) helps to transfer audio and video between browsers in real time without any downloads or plugins. Users can simply open a web page and start chatting. WebRTC enables voice, video, and screen sharing, an interactive whiteboard, and online gaming with nothing but a standard web browser.

This technology opens the door to direct and uninterrupted communication from almost any endpoint: phones, web pages, cars, and even home appliances. Its simplicity and potential ubiquity make it an excellent video streaming solution, especially in the areas of video conferencing and peer-to-peer networking.

Note that WebRTC is not a protocol, but rather a protocol stack.

WebRTC is based on three main APIs:
MediaStream

  • Allows the web browser to receive audio and video signals from cameras or from the user’s desktop
  • Allows you to access input devices such as microphones and webcams
  • Uses the user’s camera and microphone to capture and stream audio and video
  • Lets developers set limitations on audio and video streams (frame rate, video frame size, resolution, etc.)
  • Provided as part of HTML5 (RTCPeerConnection and RTCDataChannel are exclusively developed for WebRTC)

RTCPeerConnection

  • Responsible for the connection between browsers and the exchange of multimedia data received from the camera, microphone, or user’s desktop
  • Responsible for signal processing (cleaning background noise, adjusting the microphone level) and managing audio and video codecs
  • Makes it possible to send a captured live audio and video stream over the internet to another WebRTC endpoint.
  • Has functions for connecting to a remote host, maintaining and monitoring the connection, and closing the connection when it ends

RTCDataChannel

  • Sends arbitrary data
  • Each data channel is associated with an RTCPeerConnection
  • Built-in safety (DTLS) and overload control
  • Responsible for two-way data transfer over the established connection

WebRTC enables tight communication between the sender and receiver through RTCP feedback, bandwidth control, and latency targeting. Thus, it is ideal for streaming when low latency is required. Combined with RTMP to receive streaming video, this is the perfect solution for live streaming.

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VP9 video codec

Many applications are starting to use WebRTC for video streaming. As live video becomes more widespread and begins to offer higher quality, faster data transmission and reduced packet sizes are required to facilitate data transmission.

The VP9 video codec is used to compress and decompress audio and video. This helps to stream video faster and clearer. By supporting VP8, the predecessor of VP9, Safari 12.1 can share live video with other sites.

The VP9 video compression format is owned by Google and created by On2 Technologies. Its main function is to hide packet loss and clean up noisy images as well as to provide capture and playback capabilities across multiple platforms.

With VP9, users can use WebRTC to stream 720p video without packet loss or lag. It can also support 1080p video calling with the same bandwidth and helps overcome bad connections and reduce data usage to avoid costly bills for users.
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Benefits of WebRTC

Platform and device independence

For almost all major browsers, WebRTC is implemented as an open web standard that’s available as simple JavaScript APIs. In the case of mobile platforms such as Android and iOS, a special library provides access to the same functionality.

WebRTC-based browsers running on any operating system and running any web application can initiate a voice or video call with another WebRTC device or media server. This is done by integrating standard APIs from the W3C and a protocol from the Internet Engineering Task Force (IETF).With WebRTC, developers can write HTML5 code that can run on both mobile devices and desktop computers.

Secure video and voice calling

WebRTC has permanent voice and video encryption. The Secure Real Time Protocol (SRTP) is used to encrypt and verify voice and video. This is especially useful when connected to Wi-Fi and prevents the surveillance and recording of voice and video.

WebRTC and AR/VR

Creating a “social” VR experience has been a daunting task, as multi-party live video streaming has always been a major obstacle for AR/VR developers. If you need to stream video from an AR/VR-enabled device directly to a browser without adding modules or extensions, WebRTC is your only choice right now.

WebRTC supports peer-to-peer connections, opening up possibilities for multi-party conversations, broadcasting, SIP-based telephony integration, and more.

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How to choose the right WebRTC PaaS?

Your WebRTC platform as a service (PaaS) solution will become an essential communication channel for your company’s website and applications. Hence, you need to do some basic research before partnering with a supplier. In helping companies choose the right platform, we start with the following considerations.

Handling personal information

Real-time communication is governed by certain rules. By integrating WebRTC into your site, you become a telecommunications service provider according to the official definition adopted by the US Federal Communications Commission (FCC).

Some WebRTC platform providers are familiar with the specific requirements of HIPAA and other laws and regulatory frameworks. Choose a vendor that is aware of the impact of the rules and guidelines that apply to your specific industry, use case, and customer base.

Low latency

Latency means delay. Video latency is the time between capturing the feed from the source and displaying it to the end user. When you’re streaming, your users won’t technically be watching live video. They will watch with a delay.

Video latency is a critical factor for anyone looking to host high-performance live streaming, whether they’re webinar hosts, conference attendees, marketers, entrepreneurs, or gamers. If you’re streaming live video, you need to achieve low latency so your viewers can enjoy watching.

Customization and flexibility

WebRTC technology extends the capabilities of your website and applications. As such, you need to make sure you can use your PaaS SDK and APIs to include WebRTC in your website and apps in any way you like. You should be able to create the workflows, features, and UI effects you want with PaaS services.

A good PaaS vendor should offer well-documented, easy-to-use APIs and SDKs with the ability to update code as needed.

Multimedia support

Some WebRTC PaaS providers simply offer one transfer mode (video or voice). However, WebRTC allows your users to participate in video and voice communications. So why settle for just one? Give your customers the ability to communicate the way they like.

Infrastructure

Developers and engineers should be the first people you consult with before making any decision. Ask questions about your PaaS provider’s infrastructure during your business process. For example, where does the organization host its servers: internally or externally? Or are they using a trusted public cloud provider? What steps do they take to ensure uptime and scalability? How do they handle security?

Customer support

Customer support is an important factor to consider when choosing any service. Is there a phone number you can call to get help if your WebRTC platform goes down? Or will the service provider force you to wait until the appropriate industry day to identify the problem? A service provider should always have key operators on hand to help with execution and stability. This way, even a small organization can respond quickly and provide incredible help.

Scalability

After completing your research on the security and stability of an organization and its system, you should focus on the real-world features a PaaS provider offers. Start with scalability. Your customer base may grow in the medium term. So make sure the WebRTC PaaS provider you’re joining forces with offers fast service and flexibility so you can scale. Otherwise, you may find yourself in a quandary if your business grows and more customers start using your channels with advanced WebRTC features.

Pricing

Cost is obviously one of the most important factors when deciding which SDK or PaaS to use. Understanding how suppliers compare in terms of cost can be helpful in planning.

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How to implement peer-to-peer live video streaming functionality with WebRTC

We will now take a look at some streaming alternatives. Each solution is scalable, so you can stream to both large and small audiences. The multi-node WebRTC scheme involves building a network of WebRTC client viewers, each of which can stream video to three other nodes.

This way, you can broadcast the original video stream to the first peer, which can then replicate it to another three peers, which in turn can replicate it to a total of nine peers, and so on. This approach allows a scalable broadcast network to be created without the need for additional space on the service provider’s network.

Torrent-based WebRTC chains can improve performance on slow internet connections.

This approach assumes that each alternate peer can be connected to two other peers for streaming replication. Slow users can be added to such networks as independent endpoints.

Using WebRTC Relay nodes for multiple peers allows you to develop a professional-grade WebRTC streaming platform using the MediaStream and PeerConnection APIs. Along with repeaters and socket.IO for behavior control and signaling, it provides a scalable broadcast solution.

The publisher transmits the original video stream to repeaters with high download and upload bandwidth. The expander is selected to handle six outgoing streams, so six viewers can be connected to one expander. For example, such a scheme allows 24 viewers to be served with just four repeaters. By increasing the number of repeaters, you can adapt your broadcasting platform to a larger audience.

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Five apps that use WebRTC technology

1. Google Meet and Google Hangouts

Google had its own video conferencing service running in Gmail, but it required a plugin. Live video was simply not available in the browser, which is why WebRTC appeared. And it all started with a contribution from Google that is now known as webrtc.org. Today, Google Meet (or Hangouts) is a mainstream application that uses WebRTC.

2. Facebook Messenger

Here’s where Facebook has already integrated WebRTC:

  • Messenger – video chat and group video chat on mobile and browser
  • Facebook Live when co-broadcasting
  • VR Chat – video calls in Oculus
  • Workplace by Facebook
  • Instagram Live

3. Discord

Discord was originally built to serve the online gaming community and showcases how WebRTC can easily accomplish the same tasks previously handled by VoIP applications. Discord focuses on group voice calls and uses WebRTC to support in-app messaging and allows users to add a supposedly unlimited number of people to calls.

4. Amazon Chime

Amazon started using Chime for internal video conferencing and eventually made it available to customers. Interestingly, Amazon Chime started out with legacy video conferencing systems, so it is assumed they made some internal changes to the WebRTC technology for their own purposes.

In addition, Amazon uses WebRTC to support portions of its vast ecosystem of services, including Amazon Kinesis Video Streams, which is used to securely stream video to AWS, and its Alexa home assistant integration with smart home cameras and doorbells.

5. Snapchat

In early 2014, when nothing seemed to be stopping Snapchat’s explosive growth among the younger generation, the company quietly acquired AddLive, a small voice and video chat company. Snapchat almost immediately rolled out a video chat feature in its native WebRTC-based mobile app.

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Four steps to develop a video streaming app

Here are four steps to bring your video streaming app to life and simplify the development process.

1. Research your app’s target market

When developing a video streaming app, the first thing you need to consider is the target audience. Understanding your users’ needs makes development easier and leads to a better final product. Find out as much as you can about your potential users. To develop an amazing video streaming app, we suggest creating a user persona, or a detailed portrait of your ideal user.

2. Choose a monetization model

The next step is to choose the right monetization model for your app. Now that you know your users’ needs, you can predict what your users will pay for. Here are four monetization models to choose from:

  • Subscription. Users pay monthly or annually to access your service.
  • Freemium. Some content is free, and exclusive content is billed either at a pay-per-view rate or at a fixed rate for unlimited views.
  • Advertising. Ads inside the app cover expenses and generate revenue.
  • Ad blocker. Users can remove ads for a fee.

3. Hire an app development team

The third step is to find a development team you want to work with. The first phase of development will include business analysis, developing a technical specification, estimating the development cost, and planning the project. Before diving into the actual development of your video streaming app, you and your team should set up an app development workflow, choose the main features, and design a app concept.

4. Create an MVP

Creating a video streaming app is a big and complex project. We recommend launching a minimum viable product first and testing its technical and business performance. By using an MVP approach, you’ll be able to find out what users like and don’t like on your app. Then you’ll be able to consider their feedback and improve your video streaming app.

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How much does it cost to build a video streaming app?

The cost of developing a video streaming app depends on these factors:

  • Product features
  • Product design
  • Hourly rate of your development team
  • Project size and complexity
  • Technology stack
  • Number of team members on the project
  • Time frame

The biggest factor that influences a project’s cost is the hourly rate of developers, and that often depends on their location. For example, an application that costs $40,000 to build in the US will cost around $9,000 if developed by engineers in Ukraine.

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Developing your project with a company based in Eastern Europe is cost-effective and gives you access to top talent.

Quick summary of developing a video streaming app

A video streaming app, if developed correctly, can become successful and bring a lot of value to users.
Key takeaways:

  • WebRTC helps to transfer audio and video between browsers in real time without any downloads or plugins.
  • For almost all major browsers, WebRTC is implemented as an open web standard, available as simple JavaScript APIs. In the case of mobile platforms such as Android and iOS, a special library provides access to the same functionality.
  • WebRTC has permanent voice and video encryption. Secure Real Time Protocol (SRTP) is used to encrypt and verify voice and video.
  • If you are streaming live video, you need to achieve low latency so your viewers can enjoy watching.
  • When developing a video streaming app, the first thing you need to consider is the target audience.
  • Find a development team you want to work with. The first phase of development will include business analysis, developing a technical specification, estimating the development cost, and planning the project.
  • Developing your project with a company based in Eastern Europe is cost-effective and gives you access to top talent.

If you want to create a video streaming app but don’t know where to start, contact Mobindustry for a free consultation.

Mobile and Web App Development
Are you planning to expand your business online? We will translate your ideas into intelligent and powerful solutions.

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