WebRTCAlso known as Web Real-Time Communication

How WebRTC works

A page asks for the microphone with getUserMedia and creates an RTCPeerConnection. The two sides exchange SDP offers and answers through a signalling channel that WebRTC deliberately leaves unspecified; many VoIP systems use SIP over WebSocket (RFC 7118) for this. Media paths are found with ICE, STUN and TURN (see NAT traversal).

• Encryption is mandatory: all media uses SRTP keyed by DTLS; unencrypted RTP is not allowed.
• Audio codecs: Opus and G.711 are mandatory to implement (RFC 7874).
• Secure context: browsers only allow microphone access on HTTPS pages.

WebRTC and traditional SIP

A WebRTC client cannot talk directly to a plain SIP phone because of the encryption and ICE requirements. A WebRTC gateway or SBC, or a PBX such as Asterisk with WebRTC support, terminates DTLS-SRTP and ICE on one side and plain SIP/RTP on the other, transcoding Opus to G.711 where needed.

Use cases

Because WebRTC runs inside the browser, users need nothing more than a link: no softphone installation, no SIP credentials stored on their device and no firewall changes beyond HTTPS and the media ports or a TURN relay. That makes it well suited to occasional callers, visitors and staff on locked-down machines.

Click-to-call buttons on websites, browser softphones for remote agents, and in-app calling are common. A browser call can be routed like any other extension, including out through a mobile GSM gateway. GSMCalls uses Opus on the device side where supported; see the platform.