A Detailed Guide About WebRTC and WebRTC Signaling

WebRTC Signaling

Over the past few years, WebRTC technology has gained a lot of popularity. WebRTC or Web Real-Time Communication is an open-source project and specification that allows real-time communication such as voice, video, and data transfer. Communication usually takes place between the browsers and devices. With WebRTC in place, users do not require any complex plug-ins or extra hardware for communication purposes. They can easily communicate with another user from within their primary web browser. 

Benefits of using WebRTC

WebRTC brings so many benefits to the users, integrators, as well as developers that are rarely available with previous communication and collaboration platforms. Some of the key benefits of using WebRTC include:

  • Open-source API: WebRTC is an open-source application Programming Interface (API). Thus it is available for free to use. Google initially introduced it in 2011. The motive behind introducing this API was to provide a real-time, high-quality media engine that is free and available on all the major browsers.
  • Compatible with any device and platform:  WebRTC is compatible with most platforms and devices. Any WebRTC-enabled browser with any operating system and a web services application can instruct the browser to create a real-time voice or video linking to another WebRTC device or a WebRTC media server.
  • Provide secure voice and video connection: WebRTC facilitates voice and video encryption, ensuring the utmost security of exchanged information. The Secure RTP Protocol (SRTP) is utilized for encryption and validation of both voice and video. This data encryption facility proves to be highly beneficial over WiFi networks. 
  • High-quality voice and video: WebRTC generally uses the Opus audio codec that generates high fidelity voice. The Opus codec is an advanced system that is based on Skype’s SILK codec technology.
  • Reliability: WebRTC ensures a reliable session establishment. The reliable working of this API prevents server-relayed media. Thus, it decreases the latency rate and increases voice and video quality. Moreover, it helps to reduce the server load.
  • Supports multiple media streams: Another key benefit of WebRTC is that it is an adaptive network solution that can easily adjust to any network conditions. It can adjust and modify the quality of communication. 
  • Adaptive to network conditions: WebRTC is highly adaptive to the negotiation of multiple media types and endpoints. This further helps in a well-organized use of bandwidth, allowing the best possible voice and video communications. 

About WebRTC Signaling

WebRTC signaling is simply the process of creating, controlling, and dismissing a communication session. To facilitate efficient communication between two endpoints, different types of information that need to be exchanged include:

  • Session control information: To determine when to initialize, close, and modify communication sessions. This information can also be used in error reporting. 
  • Network data: To determine the location of endpoints on the Internet (IP address and port) so that callers can easily locate the person to be called.
  • Media data: To determine the codecs and media types that the callers and callees have in common. 

What is the use of signaling servers for WebRTC?

When WebRTC applications operate completely ‘in-browser, then the viewpoint of end-users matters the most. WebRTC app users just require browsers to communicate. Hence, the developers must develop server-side solutions to allow browsers to communicate with each other. In this way, the set-up of a communication platform like the OnSIP Communications Platform as a Service (CPaaS) becomes useful. 

Though WebRTC signaling servers don’t perform anything WebRTC-specific, they still play an important role in transmitting the messages around by the logic dictated by an application. WebRTC signaling allows for users to exchange metadata to synchronize communication. WebRTC uses the API RTCPeerConnection to establish peer connections and transfer audio and video media. To make a connection work perfectly, RTCPeerConnection must obtain local media conditions such as the resolution and codec capabilities for metadata. It should also collect possible network addresses for the application’s host. One thing to note here is that the signaling mechanism for transferring this crucial information back and forth is not built into the WebRTC API. 

How does WebRTC signaling work?

To connect with another user, one must know where the other person is located on the web. The IP address of a device lets internet-enabled devices send data directly between each other. The thing that is responsible for this is the RTCPeerConnection. Once the devices know how to search each other over the internet, they start exchanging information and data about which protocols and codecs each device supports. 

This way, the users are just required to exchange contact information, and the rest will be taken care of by WebRTC. This process of connecting with another user is simply known as signaling and notification. It comprises various steps, which include: 

  • Generating a list of possible candidates for a peer connection.
  • Selecting a user or an application to make a connection with.
  • The signaling part alerts another user that someone wants to connect with him. It depends on the user to accept or decline the connection. 
  • On acceptance of the connection, the first users get notified about the acceptance of the offer.
  • The first user then places the RTCPeerConnection with another user.
  • Both the users then exchange software and hardware information via the signaling server.
  • Both the users exchange location information with each other.
  • The connection either succeeds or fails.

One more important thing that one must be aware of about WebRTC is that the WebRTC specification does not encompass any standards about exchanging information. It means that the users can utilize any protocol or technology that they like.

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