A few years ago, developers used C/C++ to design software that supported video conversations. The downside to this was extended development timelines and increased expenses. However, with WebRTC, an industry-wide technology that enables real-time video communications on the web and mobile applications, the process of video conferencing for team collaborations has become much easier. By simplifying integration, reducing expenses, and ensuring browser compatibility while maintaining corporate security standards, this technology makes it easy for teams to collaborate remotely.
Video Conferencing Market Trend
Let's take a look at how the video conferencing market surpassed $15 billion USD in 2020 and is expected to grow at a 23% CAGR from 2021 to 2027.
Trend Graph for Global Market Insights:
Source: https://www.gminsights.com/industry-analysis/video-conferencing-market
What is WebRTC?
Today, real-time communication is more than just audio/video streaming. It includes cross-platform chat messages, recordings, file sharing, and screen sharing between different network configurations, making it difficult to collect and process all metrics to be transmitted to all users in a session.
WebRTC (web real-time communication) has proven to be a unified solution that provides protocols and APIs for peer-to-peer connections without needing different custom interfaces, plugins, and external integrations, making it easy to implement real-time communication between different platforms.
WebRTC requires only a URL, and all other features such as recording media, managing P2P connections, and transmitting data are handled by WebRTC protocols and APIs.
Load Testing of WebRTC-based Applications
WebRTC apps are made up of a variety of components like RTCPeerConnection, Codecs, and MediaStream. Each of these has an effect on the ecosystem as a whole. So, it is essential to consider each one while planning the performance test.
The following two helpful techniques can be used to assess the performance of WebRTC-based apps, depending on the project requirements:
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Live/Real Sessions
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As their name implies, the performance tests are carried out with actual live user sessions. This is useful when testing an application for user experience among a smaller number of users.
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Synthetic Sessions
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Synthetic Sessions uses synthetic traffic and pre-created media feeds that can be injected during user sessions. It is mostly used when working with load testing tools offering WebRTC testing services.
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Synthetic Sessions are the best options when you need to test with a large number of users concurrently in different locations, using different network environments with the goal of obtaining more actionable results that can affect the overall performance of an application.
Let's see how the load test workflow works for WebRTC-based applications:
Selecting Load Testing Tools For WebRTC
There are multiple tools available in the market that provide support for testing WebRTC-based applications. However, one should take the following criteria into consideration while selecting the WebRTC load testing tool:
- Custom media feeds of different resolutions
- Test configuration with different devices, locations, and networks.
- Live test session recording
- Simulation of multiple user sessions
- In-built APM support
Loadero and TestRTC are two popular WebRTC testing tools and below are some of their useful features:
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Loadero
- Run test on real browsers
- Support for Selenium with Java and JS
- In-built media files to simulate microphone and webcam
- Options to choose network conditioning, browser versions, computing clients, media feeds
- Provides client machine stats, browser logs, selenium logs, and WebRTC stats and data dumps
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TestRTC
- Scripting using Selenium NightWatch
- Availability of various media codecs
- Define network connection speed and characteristics
- Add customized KPIs
- Ability to run large-scale tests
- Provides metrics such as bit rate, packet loss, CPU use, and other machines metrics data
KPIs for Measuring WebRTC Performance
Since WebRTC provides a huge set of libraries to develop real-time communication applications and everyone does some level of customization, that raises the need for performance measurement of various components.
The following are some important parameters that must be measured:
- Buffering rate
- Success connections
- Average load time for starting stream in the session
- Response time for media server
- Packet loss resulting in pixelated media
- Concurrent Session
- Connection Time for peers
- Lag time for receiving the media stream
- Average Bit Rate
- Jitter
WebRTC Performance Testing Best Practices
Here are some best practices that can help to conduct the WebRTC test:
- Consider stable/unstable networks, media feeds with different formats and different session sizes during the test
- Join the test as a real user to have a real user experience
- WebRTC server components should be tuned well to establish faster communication between peers
- The performance of media and signaling servers should also be monitored
Conclusion
To get useful results from WebRTC-based applications and leverage more benefits, performance testing should be performed using both live sessions that provide real user-based experience, as well as predicted synthetic data that would give better insights for session quality specific to networks and devices. To know more about performance testing services, contact QASource experts now.
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