WebRTC is the open, browser-native standard for real-time audio, video, and data communication, jointly maintained by the W3C (the JavaScript APIs) and the IETF (the network protocols), and shipped in every major browser and mobile platform. Rather than a single feature, it is a bundle: media capture from cameras and microphones, audio and video codecs, congestion control that adapts to changing network conditions, and mandatory encryption of the media (DTLS-SRTP) that cannot be turned off. Because all of this is built into the platform and always encrypted, WebRTC is the default transport for live telemedicine consults — there is no plugin to install and no unencrypted mode to misconfigure.
The always-on media encryption is also why WebRTC aligns cleanly with HIPAA's transmission-security expectations under the Security Rule (45 CFR §164.312): every audio and video packet is encrypted on the wire by default.
What WebRTC deliberately leaves out is just as important as what it includes, and it is where telemedicine platforms actually differentiate. The standard says nothing about signaling (how two parties find each other and agree to connect), identity (who the participants really are), recording, or multi-party topology (how a group call is routed). A product team must supply all of that: a signaling service, authentication tied to the patient and clinician, a recording pipeline with consent and retention, and a media server such as an SFU for group consults. The common pitfall is assuming WebRTC "handles security" end to end — it secures each hop, but the moment media passes through a server, that server can see it unless you add true end-to-end encryption (E2EE) on top.
