WebRTC can be seen as an extension of VoIP, bringing the capabilities of making phone calls, video calls, text chat, and even Peer 2 Peer file transfers, directly through your web browser or mobile apps. WebRTC is a JavaScript API developed with the purpose of establishing communication functionality directly in your web browser without the need for downloading any plugins or clients. With WebRTC you can make phone or video calls straight from your web browser without having to load any plug-ins, or download a stand-alone softphone or web conferencing app.

WebRTC works on all popular mobile devices, and on the most widely used web browsers like Safari, Microsoft Edge, Mozilla Firefox, Google Chrome, and Opera. With WebRTC, it’s possible to build your own communication apps or platforms or embed them directly into your webpage. With WebRTC you can open up a web page and start a video chat without downloading any plugins.

WebRTC uses the Session Description Protocol (SDP) to describe the parameters of peer-to-peer connections. SDP does not deliver any media itself. Instead, it is used to describe the “session profile,” which represents a list of properties of the connection – that is the types of media to be exchanged (audio, video, and application data), network transports, used codecs, and their settings, bandwidth information, and other metadata.

WebRTC can be considered an extension of VoIP that brings the potential of making phone and video calls, chats, and Peer 2 Peer file transfers directly to the web browser or mobile app. WebRTC allows for more widespread adoption of VoIP, acting as an extension for VoIP into new platforms. Browser-based calling also reduces security risks for enterprises.

WebRTC architecture is designed to provide the best communication experience with maximum security. It mandates best security practices (like WSS and DTLS-SRTP) and supports high definition codes (such as OPUS and VP9) that guarantee the best communication experiences.

Further, WebRTC softphones are designed to be resilient to network hiccups, ensuring great calling experiences irrespective of network stability.

VoIP telephony based on the browser removes the need for physical and softphone equipment altogether. Compared to other VoIP devices, this solution has several advantages – both hardware and software.

A browser-based solution using WebRTC is an even more elegant solution. For web-based calling, it doesn’t matter what hardware or operating system you are using. The solution will operate in any WebRTC-supporting browser. There is no need to bother with updates either.

Hackers abound on the internet. However, WebRTC has a few inherent architectural advantages that make it difficult for hackers to break into:

• Peer to Peer (P2P):  After the Peer Connection has been established, all video, audio, and data traffic over WebRTC are sent directly from one peer to another. This means there is no intermediary server in which the conversation can be hacked. This is inherently more secure than any application that uses a media server in the middle. Any P2P WebRTC call cannot be listened to by a third party.

• Encrypted: WebRTC encrypts all data while it’s in transit. This means even if someone is watching your network traffic, they will not be able to decrypt the video or audio being sent between two peers and listen in to your conversation.

• Data Channel: In addition to video and audio, WebRTC also offers a Data Channel layer with the same Peer-to-Peer encryption. This means your application can use the Data Channel to securely send files between two peers without fear of someone being able to intercept that data.

WebRTC and SIP serve different purposes, with WebRTC better suited for modern web-based communication. It enables peer-to-peer connections, advanced codecs, and browser-native functionality without plugins, simplifying development. SIP requires more infrastructure, while WebRTC works directly with JavaScript APIs. WebRTC also ensures better security with mandatory encryption and NAT traversal.

WebRTC and VoIP are related but different. VoIP transmits voice over the internet, while WebRTC is a specific technology enabling real-time communication in browsers without plugins. WebRTC supports VoIP and features like peer-to-peer data transfer and screen sharing.

Yes, WebRTC can use SIP as a signaling mechanism to establish and manage VoIP calls. However, WebRTC itself is a flexible framework and doesn’t require SIP, allowing it to work with other protocols too. Compatibility depends on the specific softphone implementation.