WebRTC Interview Questions & Answers

  1. Question 1. What Is Webrtc?

    Answer :

    WebRTC is an open framework for the web that enables Real Time Communications in the browser. It includes the fundamental building blocks for high-quality communications on the web, such as network, audio and video components used in voice and video chat applications.

    These components, when implemented in a browser, can be accessed through a JavaScript API, enabling developers to easily implement their own RTC web app.

    The WebRTC effort is being standardized on an API level at the W3C and at the protocol level at the IETF.

  2. Question 2. Why Should I Use Webrtc?

    Answer :

    We think you’ll want to build your next video-chat style application using WebRTC. Here’s why:

    • A key factor in the success of the web is that its core technologies – such as HTML, HTTP, and TCP/IP – are open and freely implementable. Currently, there is no free, high-quality, complete solution available that enables communication in the browser. WebRTC enables this.
    • Already integrated with best-of-breed voice and video engines that have been deployed on millions of endpoints over the last 8+ years. Google does not charge royalties for WebRTC.
    • Includes and abstracts key NAT and firewall traversal technology, using STUN, ICE, TURN, RTP-over-TCP and support for proxies.
    • Builds on the strength of the web browser: WebRTC abstracts signaling by offering a signaling state machine that maps directly to PeerConnection. Web developers can therefore choose the protocol of choice for their usage scenario (for example, but not limited to, SIP, XMPP/Jingle, et al.).
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  4. Question 3. What Is The Opus Audio Codec?

    Answer :

    Opus is a royalty-free audio codec defined by IETF RFC 6176. It supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s, frame sizes from 2.5 ms to 60 ms, and various sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, where the entire hearing range of the human auditory system can be reproduced).

  5. Question 4. What Is The Isac Audio Codec?

    Answer :

    iSAC is a robust, bandwidth-adaptive, wideband and super-wideband voice codec developed by Global IP Solutions, and is used in many Voice over IP (VoIP) and streaming audio applications. iSAC is used by industry leaders in hundreds of millions of VoIP endpoints. This codec is included as part of the WebRTC project.

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  7. Question 5. What Is The Ilbc Audio Codec?

    Answer :

    iLBC is a free narrowband voice codec that was developed by Global IP Solutions, and is used in many Voice over IP (VoIP) and streaming audio applications. In 2004, the final IETF RFC versions of the iLBC codec specification and the iLBC RTP Profile draft became available. This codec is included as part of the WebRTC project.

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  9. Question 6. What Is The Vp8 Video Codec?

    Answer :

    VP8 is a highly-efficient video compression technology developed by the WebM Project. It is the video codec included with WebRTC.

  10. Question 7. What Other Components Are Included In The Webrtc Package?

    Answer :

    Audio: WebRTC offers a complete stack for voice communications. It includes not only the necessary codecs, but other components necessary to great user experiences. This includes software-based acoustic echo cancellation (AEC), automatic gain control (AGC), noise reduction, noise suppression, and hardware access and control across multiple platforms.

    Video: The WebRTC project builds on the VP8 codec, introduced in 2010 as part of the WebM Project. It includes components to conceal packet loss and clean up noisy images, as well as capture and playback capabilities across multiple platforms.

    Network: Dynamic jitter buffers and error concealment techniques are included for audio and video, which help mitigate the effects of packet loss and unreliable networks. Also included are components for establishing a peer-to-peer connection using ICE / STUN / Turn / RTP-over-TCP and support for proxies.

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  12. Question 8. Are Webrtc Components Subject To Change?

    Answer :

    WebRTC is based on a API that is still under development through efforts at WHATWG, W3C and IETF. We hope to get to a stable API once a few browser vendors have implementations ready for testing. Once the API is stable, our goal will be to offer backwards compatibility and interoperability. The WebRTC API layer will be our main focus for stability and interoperability. The components under it may be modified to improve quality, performance and feature set.

  13. Question 9. Do I Have To Be A Programmer To Use Webrtc?

    Answer :

    Yes, to build WebRTC support into a software application or contribute improvements, programming skills are required. However, usage of the JavaScript APIs that call WebRTC in the browsers will only require typical web development skills.

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  15. Question 10. Is The Webrtc Project Owned By Google Or Is It Independent?

    Answer :

    WebRTC is an open-source project supported by Google, Mozilla and Opera. The API and underlying protocols are being developed jointly at the W3C and IETF.

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  17. Question 11. Are The Webrtc Components From Google’s Acquisition Of Global Ip Solutions?

    Answer :

    Yes. Some components, such as VoiceEngine, VideoEngine, NetEQ, AEC, et al. all stem from the GIPS acquisition.

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  19. Question 12. What Codecs Are Supported In Webrtc?

    Answer :

    The currently supported voice codecs are G.711, G.722, iLBC, and iSAC, and VP8 is the supported video codec. The list of supported codecs may change in the future.

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  21. Question 13. Please Explain How Webrtc Is Free Of Charge?

    Answer :

    Some software frameworks, voice and video codecs require end-users, distributors and manufacturers to pay patent royalties to use the intellectual property within the software technology and/or codec. Google is not charging royalties for WebRTC and its components including the codecs it supports (VP8 for video and iSAC and iLBC for audio). 

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  23. Question 14. What Does This License Let Me Do?

    Answer :

    Like most BSD licenses, this license allows you to use the WebRTC code with a minimum of restrictions on your use. You can use the code in proprietary software as well as open source software.

  24. Question 15. Do I Need To Release The Source If I Make Changes?

    Answer :

    No, the license does not require you to release source if you make changes. However, we would love to see any changes you make and possibly incorporate them, so if you want to participate please visit the code review page and submit some patches.

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  26. Question 16. Why Is There A Separate Patent Grant?

    Answer :

    In order to decouple patents from copyright, thus preserving the pure BSD nature of the copyright license, the license and the patent grant are separate. This means we are using a standard (BSD) open source copyright license, and the patent grant can exist on its own. This makes WebRTC compatible with all major license scenarios.

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  28. Question 17. What If Someone Gets The Code From Google And Gives It To Me Without Changes. Do I Have A Patent Grant From Google?

    Answer :

    Yes, you still have the right to redistribute and you still have a patent license for Google’s patents that cover the code that Google released.

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  30. Question 18. What If Someone Makes A Change To The Code And Gives It To Me. Do I Have A Patent License From Google For That Change?

    Answer :

    You still have the right to redistribute but no patent license for the changes (if there are any patents covering it). We can’t give patent licenses for changes people make after we distribute the code, as we have no way to predict what those changes will be. Other common licenses take the same approach, including the Apache license.

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  32. Question 19. What If Google Receives Or Buys A Patent That Covers The Code I Receive Sometime After I Receive The Code. Do I Have A Patent Grant For That Patent?

    Answer :

    Yes, you still have the right to redistribute and you still have a patent license for Google’s patents that cover the code that Google released.

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  34. Question 20. What If My Competitor Uses The Code And Brings Patent Litigation Against Me For Something Unrelated To The Code. Does He Or She Still Have A Patent License?

    Answer :

    Yes, he/she still has the right to redistribute and he/she still has a patent license for Google’s patents that cover the code that Google released.

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  36. Question 21. What Is Full Form Of Webrtc?

    Answer :

    Web Real-Time Communication.

  37. Question 22. What Is Websocket?

    Answer :

    WebSocket is a protocol providing full-duplex communication channels over a single TCP connection.

    OR

    WebSocket can do communication between client to server OR server to client.

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  39. Question 23. What Is Whatwg?

    Answer :

    WHATWG is a community of people interested in evolving HTML and HTML related technologies. 

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  41. Question 24. What Is Full Form Of Whatwg?

    Answer :

    Web Hypertext Application Technology Working Group.

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  43. Question 25. Where Will Webrtc Effect?

    Answer :

    Real time communication such as text/audio/video between users by utilizing the browsers.

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  45. Question 26. What Are Manjor Components In Webrtc?

    Answer :

    1. Audio Calls
    2. Video Calls
    3. Share data viz peer-to-peer
    4. Access Camera and Microphone
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  47. Question 27. Who Started On Webrtc? And When?

    Answer :

    Google on May 2011. 

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  49. Question 28. What Is A Vp8?

    Answer :

    VP8 (Video Compression Format or Video Compression Specification) is a specification for encoding and decoding high definition video as either a file or a bitstream for viewing. The VP8 codec is – in contrast to the H.264 codec – free since Google has released all of the VP8 patents it owns under a royalty- free public license.

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  51. Question 29. Is Webrtc Secure?

    Answer :

    WebRTC is an open-source technology available for free on any Web browser and it is plug-in free. As such WebRTC users fear hackers might be able to listen in on conferences, access user data, or even private networks.

    As with any file or software you may download from the internet, it’s extremely dangerous if your PC doesn’t have secure firewall configurations. The same applies for downloading VoIP applications. With WebRTC however there’s no need to worry about this, as you don’t need to install any plugins or clients on your PC.

    Security and encryption is not an optional WebRTC feature, as it has native built-in features that address security concerns. What’s more, WebRTC offers end-to-end encryption between peers on almost any server ensuring safe, private and secure real-time communications.

    WebRTC requires the user to explicitly allow access to his camera and microphone. This ensures that the user is made aware that his camera and microphone will be turned on. When the user allows access, a red dot will appear on that tab, providing a clear indication to the user, that the tab has media access.

    Prior to using WebRTC, users are notified that a specific website is trying to access their camera and microphone. If a tab has access to their media devices, users are also notified by the browser with a blinking red spot on the tab.

    For WebRTC to transfer real time data, the data is first encrypted using the DTLS (Datagram Transport Layer Security) method. This is a protocol built into all the WebRTC supported browsers from the start (Chrome, Firefox and Opera). On a DTLS encrypted connection, eavesdropping and information tampering cannot take place.

    Other than DTLS, WebRTC also encrypts video and audio data via the SRTP (Secure Real-Time Protocol) method ensuring that IP communications – your voice and video traffic – can not be heard or seen by unauthorised parties.

    Just like anything you do online, whether downloading a VoIP application like Skype, downloading a movie, or even transferring information and files via email, there is a risk of malicious intrusions. However, WebRTC technology safeguards the transmission of sensitive data through the standards explained above, ensuring secure real time communications.

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  53. Question 30. What Is Web Conferencing?

    Answer :

    Web Conferencing is an online service by which you can hold live meetings, conferencing, presentations and trainings via the internet particularly on TCP/IP connections. You can connect to the conference either by telephone or using your computer’s speakers and microphone through a VoIP connection.

    Web conferencing usually allows real time point-to-point communication as well as multi task communications from one sender to many receivers in separate locations. Depending on the service, either an application (additional software) is downloaded and installed or a web-based application is launched in the attendee’s browser.