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WebRTC and SIP over WebSocket

How a browser reaches SIP: RFC 7118 on the signaling socket, ICE for candidates, and DTLS-SRTP for the media.

5 rooms30 quiz items

Rooms

  1. Signaling, JSEP, and the media plane

    WebRTC names a browser protocol suite. JSEP leaves the signaling protocol to the application. SIP over WebSocket is one such protocol.

  2. RFC 7118: the sip subprotocol

    The handshake must offer sip. Via uses WS or WSS. The URI transport parameter added by the RFC is ws.

  3. ICE, STUN, and TURN

    ICE gathers host, server-reflexive, and relayed candidates. STUN and TURN are the tools. The default STUN port is 3478.

  4. DTLS-SRTP and the required audio codecs

    WebRTC media is SRTP. The profile is RTP/SAVPF. Opus, PCMA, and PCMU are required audio codecs.

  5. Browser softphone call flow

    A study ladder: WebSocket handshake, SIP INVITE, answer, then DTLS on the media path. Step the hops.

Ecosystem

SIP does not replace workforce management, quality monitoring, CRM, emergency policy, or a messaging registry. It is the session seam those systems hang from. Compliance notes on this page are educational and are not legal advice.

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