WebRTC Protocol
WebRTC is a highly flexible streaming protocol suitable for nearly every audio/video
use case. While this characteristic and its media encryption capabilities make it a
popular choice among broadcasters, this protocol does have three major drawbacks
potential streamers should be aware of. Read on to learn what they are, how much
they might impact your stream, and learn if WebRTC is right for your project.
What Is WebRTC?
Web Real-Time Communications (WebRTC) is an open-source communications
protocol that enables real-time voice, text, and video streaming between web
browsers and devices. With the help of signaling servers, WebRTC is able to manage
multiple device connections and ensure their integrity.
WebRTC provides software developers with application programming
interfaces (APIs) written in JavaScript. Developers can leverage APIs to create a peer-
to-peer connection (P2P) between internet web browsers and mobile applications
without worrying about compatibility and support for multimedia content.
Data transfers occur in real-time without needing custom interfaces, extra plugins, or
additional software -- WebRTC enables real-time audio and video streaming simply
by opening a webpage. However, to access more advanced features like screen
sharing, you can either install a plug-in or build your own.
How Does WebRTC Work?
WebRTC embeds real-time communications technologies within web browsers using
JavaScript, APIs, and HTML. It makes audio, data, and video streaming between most
major web browsers user-friendly and easy to implement. Before audio and video
files are sent, they must be compressed due to their large size. Also, media received
over a peer-to-peer connection must be decompressed. WebRTC uses a codec
process to do this.
WebRTC typically connects users by transferring real-time audio, video streams, and
data from device to device using P2P. But, if users are on different Internet Protocol
(IP) networks with Network Address Translation (NAT) firewalls that prevent real-time
communication (RTC), Session Traversal Utilities for NAT (STUN) servers can be used
to translate an IP address into a public internet address, so P2P connections can be
established. WebRTC APIs initiate and monitor P2P connections between devices via
browsers and facilitate bidirectional data transfer over multiple channels.
Why Choose WebRTC?
WebRTC has a wide range of benefits for both the broadcaster and the receiving
audience. Here are the top three:
1. WebRTC eliminates much of the in-house manual integration work required of
IT teams by adjusting communication quality, bandwidth, and traffic flow
whenever network conditions change.
2. WebRTC is supported by most major web browsers, including Google Chrome
for desktop and Android, Mozilla Firefox for desktop and Android, and Safari,
and works on any operating system.
3. WebRTC does not require third-party components or plugins and is free
open-source software.
When To Use WebRTC
Use WebRTC to facilitate real-time communications over the internet for a number of
use cases:
Businesses can host video chats and meetings on video calling platforms, such
as Zoom, Microsoft Teams, Slack, or Google Meet.
Telehealth organizations can connect patients and providers securely over a
web browser for virtual appointments.
Home and business surveillance companies can use WebRTC as a connecting
agent between browsers and security cameras.
Sporting, news, and other live events can use WebRTC to deliver a high-
quality stream in real time.
Students and teachers can facilitate remote learning over WebRTC.
When Not To Use WebRTC
While WebRTC is encrypted and its APIs require secure origins via HTTPS, there are
still some concerns around security questions developers must consider. There are
no signaling processing methods specified for WebRTC signaling. If you do not wish
to decide which security protocols to use and are unsure if the protocols you select
can be maintained with WebRTC, you might consider looking into an
alternative video streaming protocol.
While security is a top concern among many potential WebRTC users, here are three
other aspects of the protocol worth considering where they fall on your list of
development priorities:
1. Bandwidth: Each user must establish a P2P browser connection, making
bandwidth an issue.
2. Cost: Maintenance costs can be high because WebRTC requires powerful
servers.
3. Service Standards: There is no definitive quality of service standards, which
means that the quality of video or audio over the internet may be
inconsistent.
Frequently Asked Questions
Does Zoom use WebRTC?
No, Zoom uses a custom protocol solution with levels of fallback in the event their
custom protocol is unavailable.
Is WebRTC owned by Google?
Any one company does not exclusively own WebRTC. Google developed WebRTC
and has worked with W3C, Mozilla, and Ericsson Labs to continue to iterate and
improve the protocol.
Is WebRTC only for browsers?
Yes, WebRTC is available to all modern browsers and will not work on mobile iOS
apps out of the box.
Does WebRTC need a server?
Yes, WebRTC requires a signaling server for devices that wish to communicate to
locate one another.
What is the difference between Websocket and WebRTC?
WebRTC is intended to facilitate real-time communication, mostly voice and video,
over browsers, while Websockets are intended to help browsers and web servers
communicate.
Can I use WebRTC for live streaming?
Yes, WebRTC is a popular choice for live streaming because it is a free, open-source
protocol compatible with all major browsers and requires little additional
engineering.
Is WebRTC a TCP or UDP?
WebRTC is a unique browser protocol because it transmits its data over UDP instead
of TCP, like most others.
What is WebRTC network limiter?
WebRTC offers a network limiter extension for most browsers that enables the user
to decide how the WebRTC network traffic is routed. It can also make private IP
addresses invisible to the public internet.
What is a WebRTC gateway?
A WebRTC gateway bridges the Web and traditional telecom worlds, providing
signaling interworking, media interworking and transcoding, and application
interworking functions. The WebRTC gateway can be a turn-key appliance or a
software-based solution that runs industry-standard servers.
What is a data channel?
A data channel is the path transmitted data takes when transferred from one device
to another.
Now that you know what WebRTC is, and how it works, you might want to check out
our WebRTC tutorials and beginner resources to refer back to if you have any
questions or want to brush up on a particular facet of the protocol.
