VoIP and WebRTC: Real-Time Voice and Video on the Web VoIP and WebRTC bring real-time voice and video to the web. VoIP is the broad idea of sending voice over the internet, while WebRTC is a set of browser APIs that lets apps capture media, create connections, and stream audio and video directly in a page. This pairing makes it easier to add calling features to websites or apps, from a simple one-on-one call to a small conference. ...
VoIP and WebRTC: Real-Time Communication Over Networks Real-time communication lets people talk, share, and collaborate over the internet. VoIP and WebRTC are two common paths to this goal. VoIP has a long history with signaling and voice networks, while WebRTC brings audio, video, and data into the browser. Together, they cover desk phones, smartphones, and desktop apps. Both approaches separate signaling from media. Signaling coordinates who talks to whom and when, while media transport carries the actual sound and video. In practice, you might use SIP for signaling in VoIP setups or use WebRTC’s built-in negotiation in browser apps. WebRTC also includes data channels for messages or file transfer. ...
VoIP and WebRTC: Modern Voice Communication VoIP and WebRTC are changing how we talk. VoIP is the broad idea of sending voice over the Internet, used by business phone systems and cloud services. WebRTC is a set of browser technologies that lets people talk, chat, and share in real time without plugins. Together they power daily communication—from a team standup to a customer chat. What VoIP delivers VoIP lowers costs, supports remote work, and scales with your needs. It lets calls run over data networks, not traditional phone lines. You can route calls through a cloud PBX, a SIP trunk, or a hybrid setup. Typical signaling uses SIP, while media is carried by RTP. This separation makes it easier to add features like voicemail, call recording, and auto attendants. ...