An Intro to WebRTC
WebRTC is technology that allows browsers and other devices to interact and communicate with each other in real-time. Browsers using WebRTC do not need extra plug-ins or downloads -- it all works natively in the browser.
Using Bandwidth's APIs, you can write audio and video enabled applications that run in browsers or mobile devices, and connect phone calls and real-time video sessions between these devices.
Part of the Magic of WebRTC was the decision to embed a standardized real-time media handling capability into all browsers, allowing real time communications between any collection of desktop and mobile browsers using a globally adopted standard.
Standard browser APIs for access to sources of media ( getUserMedia for camera and microphone, getDisplayMedia for screen content) and The RTCPeerConnection API for controlling media flows to remote endpoints enable all parties to exchange real time media in an efficient and uniform manner.
One of the intended side-effects of the browser-centric approach taken with WebRTC is the broad spectrum of media that can be exchanged, and the flexibility available in exchanging that media. The endpoints of any media streams are free to adapt to the constraints of the channel with a suite of codecs, and additional application-specific data can be exchanged using WebRTC data channels.
Whether it is voice, video, screen-sharing or application-specific data, WebRTC supports it.
Notice that nothing above mentions finding / addressing the other participants in a communication, and nothing above mentions control at the session level for adding and removing participants in a session. That is because the browser WebRTC infrastructure leaves all of that to the controlling application. WebRTC is explicitly unopinionated on that front.
Although this sounds like a deficiency it results in flexibility in creation of applications and communication patterns that ideally suit the creator's intended use.
Your application completely controls who "talks to" whom.
Bandwidth enhances the use of WebRTC by wrapping standardized multimedia capabilities in an API and SDK suite that simplifies the creation of communication applications that fully leverage the power of both WebRTC and the Bandwidth voice network.
At a very high level the model simplifies the browser and server-level interactions with the media and signaling/session control levels:
This approach allows for simple and powerful...
- coordination of browser and web app for session participation
- find-grained control of all aspects of the communication, and
- controlled interaction with the Programmable Voice capability set
Secure Coordination of Server, Browser, and Media
In the Bandwidth WebRTC model all media flows are mediated by Bandwidth Media Control Units (MCUs and FCUs). This allows application control over the media characteristics of the communication, and facilitates interaction with the Bandwidth Programmable Voice network.
If you create an application to do some cool WebRTC stuff, one of the implied requirements is security: the media sessions that are candidates to participate in a session should be secured prior to admission. In the Bandwidth solution a JWT is issued for every participant in a session, and that JWT is used in establishing media connections to the Media Server infrastructure, and thus to the other participants in the communication. These secure tokens are minted and issued by the server application logic, and used to coordinate with media streams that are admitted into the communication.
Secure Server to Server interaction provides secure access to the Bandwidth WebRTC and Programmable voice capabilities, and tokens are exchanged with the Browser component to allow controlled and secure access to media flows.
Fine Grained Control: Application Flexibility
The model underlying Bandwidth's WebRTC solution allows the application to control the flows of media with exquisite precision. There are 4 levels of control, in increasing levels of precision...
- sessions - groups of users involved in a communication
- participants - the addressable members that are involved in a session
- subscriptions - essentially who "listens to" whom. It is possible to create complex topologies to address specific application needs
- media - the specific media that is exchanged between any subscribed participants and/or sessions
With these elements under application control almost any real time communication imaginable can be created.
The details of this model and its control are covered in greater detail in the document section Bandwidth WebRTC Model Details
But that's not all... :-)
Bandwidth brings a comprehensive Programmable Voice network capability to the overall equation, and the Bandwidth WebRTC capabilities are designed to integrate with these capabilities in a way that enhances both the WebRTC and the Programmable Voice capabilities. The customer application can leverage both capability sets in an integrated manner.
This enables creation of Browser-based endpoints that participate in voice network, WebRTC video communication that extends into the PSTN, and everything in between.
The Programmable Voice infrastructure uses SIP URI and User-User information to place, transfer, bridge, or conference WebRTC endpoints into PSTN calls. On the WebRTC side of the equation, the interconnection with the Programmable Voice shows up as a WebRTC participant in the overall WebRTC session, managed with the same level of fine-grained control described above.
Learning about WebRTC
The internet is your friend, and there is lots of information out there.
A couple of valuable sites are...
- The Mozilla MDN for browser details
- The W3C
- An industry perspective