What is WebRTC and how to make online communication and streaming more comfortable?
Interactive platforms (such as blogging, online education, gaming and spectator enthusiasts) play a crucial role in broadcasting live as in such platforms it is possible to exchange views in real-time and, therefore, make important decisions at that moment. Regardless of the profitability of residing stream services, there are also benefits to all who participate in such broadcasts, for which the main factor for the streaming service providers is the viewers and authors themselves. Thus, the channel should fulfill the needs of both sides of the traffic to prevent that they change to another platform.
What is WebRTC?
WebRTC (Web Real Time Communication) is an open project that offers web browsers and mobile applications with real-time communication by providing easy-to-use APIs. It enables audio-video interaction to work on pages of websites by permitting direct peer-to-peer communication helping to avoid timely installing plugins or downloading apps. Supported by Google, Microsoft, Mozilla, and Opera W3C (W3C) and the Internet Engineering Task Force (IETF) are spearheading the standardization of this technology.
The operation of WebRTC that negotiorates connections and exchanges data between peers is fascinating in terms of its functionality. When setting up a WebRTC campaign, one client needs to pass an offer to another client, representing the way of encoding media streams. This negotiation is achieved through the SDP protocol. WebRTC activates communication between peers using Interactive Connectivity Establishment (ICE) which could involve going around firewalls and Network Address Translators (NATs). Additionally, ICE combines STUN(Session Traversal Utilities for NAT) servers to find out each peer’s public IP address and employs TURN(Traversal Using Relays around NAT) servers as a back-up if the direct peer-to-peer connection fails.
In a few words, WebRTC allows a peer-to-peer connection between users that is very advantageous over other communication channels because the latency and quality of video and audio transmissions are better. The simplicity, alongside the low delay and real-time communication provided by WebRTC, makes the technology very attractive for real-time communication on the web, which makes it a technology for rich, high-quality, browser, mobile platform and IoT device development via a common set of protocols for their communication.
Stay in the Loop with WebRTC Insights!
Subscribe now to receive the latest industry news and stay informed.
WebRTC is an open standard, which makes it an independent technology and enables it to be implemented into the platform, which is used for streaming, for example. Nevertheless, a way to deal with integration of WebRTC that really works effectively is to use a ready-made infrastructure that is trusted and proven. Flussonic Media Server is an example of an application that has long been using WebRTC platform as the technology basis of their product. We provided a unique sourcing technology and, if required, even a little bit more, so that platforms can manage the necessary use cases and covering real life situations of their authors and viewers.
By incorporating Flussonic Media Server for one-to-many WebRTC communication on your streaming platform, you can reap several benefits that enhance the live streaming experience by providing a greater sense of immersion and interactivity.
- Security. Data security involves using the browser to publish the content directly to a website because you don’t have to share your data with the streaming apps.
- Stable (consistent) low latency is, however, a prerequisite for timely feedback from the author to the viewers (and vice versa) which includes a healthy sensation of communication.
- Promise of best sound quality in minimal latency, sufficient to use for phone calls, guarantees excellent audio communication and instant reaction in real-time streaming.
- The adoption of WebRTC makes possible significantly lowering the latency for iOS device users “to sub-second values”. iOS does not support MSE, and thus the only way to get video with low latency remains WebRTC. This is particularly critical with regard to live broadcasting, since delay can impact the way content is perceived.
In addition, the Flussonic Media Server:
- Ensures correct load balancing by considering node usage, which helps to disperse the stream to an appropriate server and as a result improves hardware resource utilization.
- Flexibly switches between TCP and UDP and to select the most suitable protocol for data transmission depending on the type of data and network conditions.
- Constantly monitors the available bandwidth and enables quick reaction on the transmission quality drops.
- Offers great scalability to cover such a need due to countless, millions, and even more viewers.
- Supports WebRTC ABR: for example, some viewers may see quality in low resolution (their Internet connection is not capable of holding more), and in contrast some others will be fine on the maximum possible bitrate.
When to Choose Flussonic with WebRTC for Your Live Streaming Needs?
We have various periodical and push streaming platforms, e.g. Twitch, Periscope, Zoom, YouTube, and Facebook, that enable content creators and their followers to interact as they live stream. In contrast, the platforms can sometimes be constrained by T&C and inadequate technologies in meeting certain standards.
For example, in some countries, Zoom is either blocked or has some limitations for its use in some organizations, especially among those dealing with the government-linked entities. Trending services like Twitch and YouTube may ban the content that isn’t suitable for broadcasting, like adult streaming for example. The video will be taken down in advance. Debates over data security are unlikely to be on the public agenda during scheduled sessions with a psychologist or virtual medical consultation that parents have arranged. Firms in the business of conduct live seminars may also be interested in protecting their material. The noticeable thing about YouTube copyright policy is that it is very much strict, and it can remove videos anytime violating these rules. If it happens, it can involve disabling the author’s channel.
As for the technologies, platforms use generally popular standards such as MPEG-DASH, HLS and RTMP, but they have a disadvantage of low sound quality due to frequency bands prone to distortions, do not support instability of the network connection and not all of the protocols support real-time broadcast with the sub-second latency on iOS. On the other hand, there are applications like YouTube, Periscope, and Facebook which have to be installed first prior to streaming, not forgetting that some authors and viewers may never be interested in that as well.
Weak points of WebRTC
However, while WebRTC possesses many important merits, there are instances where it makes more sense to go with TCP transmission. undefined
WebRTC uses it as protocol. It simplifies communications but on other side it erodes the security features. To recall, TCP is a series of steps which takes place starting from the time the client starts connection up to the point where the other end confirms that the message was transmitted properly. In practice, availability is guaranteed by the principle of being able to verify at any way to the other party has received the data or not. However, in contrast with UDP, which sends data in a seamless manner, it is not possible to verify the fact that the other side really received the data.
All WebRTC implementations are not compatible with each other, since the standard only describes methods for sound and video transmission, leaving more complex features such as messaging, file transfer, user management, and other constructs out of scope. Meaning that can not connect from one WebRTC app to another.
WebRTC employs the mechanism of the real IP addresses of users. Also, both proxies or Tor network will be unable to secure your anonymity neither. These two kind of services will help you mask your IP address.
Summary
Although not all live events will require WebRTC technology, but if interactivity and audience engagement are the central elements of your business, and if real-time responsiveness, high-quality audio, adapting for different types of internet connections, or restrictions from using other platforms, then building your live streaming solution using WebRTC is the right way. Flussonic Media Server has all the benefits of WebRTC technology without creating a new infrastructure from a scratch. It is a ready to use platform that has you to focusing on your business, content monetization, contracting writers together with viewers who need live experience, while we solely are occupied of stream technology.