# What is WebRTC?

> WebRTC (Web Real-Time Communication) is an open standard that lets browsers and apps exchange real-time audio, video, and data peer-to-peer without a plugin. It bundles media capture, the Opus and VP8/VP9 codecs, encrypted transport (SRTP), and NAT traversal (ICE/STUN/TURN), negotiating sessions with SDP. WebRTC powers in-browser click-to-call, video, and voice-agent experiences, often bridged to SIP and the PSTN by a gateway. See /answers/what-is-sip, /answers/what-is-srtp, and /answers/what-is-nat-traversal.

- Canonical URL: https://www.broadnet.me/answers/what-is-webrtc
- Language: en
- Last reviewed: 2026-07-22
- Publisher: BroadNet (https://www.broadnet.me)

## Related

- [All answers](https://www.broadnet.me/answers)
- [Glossary](https://www.broadnet.me/answers)
- [What is SIP (Session Initiation Protocol)?](https://www.broadnet.me/answers/what-is-sip)
- [What is SRTP (Secure RTP)?](https://www.broadnet.me/answers/what-is-srtp)
- [What is NAT traversal in VoIP?](https://www.broadnet.me/answers/what-is-nat-traversal)
- [What are voice codecs (G.711, Opus)?](https://www.broadnet.me/answers/what-is-voice-codec)
- [How does a voice API and SIP trunking work?](https://www.broadnet.me/answers/voice-api-sip-trunking)
- [Voice](https://www.broadnet.me/products/voice)
- [Voice integration](https://www.broadnet.me/products/voice)