VoIP stands for Voice over IP.
VoIP is an umbrella term for all voice and video real-time communications over the internet. In that sense, WebRTC is a subset of VoIP.
VoIP and WebRTC
Traditional VoIP systems rely on protocols like SIP for signaling and RTP for media transport, requiring dedicated softphones or hardware IP phones. WebRTC brought VoIP capabilities directly into the browser with no plugins required, dramatically lowering the barrier to entry.
Key differences between traditional VoIP and WebRTC:
- Encryption: WebRTC mandates encryption (SRTP + DTLS) for all media. Traditional VoIP often sends media unencrypted
- NAT traversal: WebRTC has ICE built in. Traditional VoIP relies on external solutions like SIP ALGs
- Codecs: WebRTC uses modern codecs like Opus. Traditional VoIP commonly uses G.711
- Signaling: WebRTC leaves signaling to the application. Traditional VoIP standardizes on SIP or H.323
Many CPaaS vendors provide bridges between WebRTC and traditional VoIP infrastructure, enabling interoperability between web-based and legacy phone systems.
