Figure 2.5, “WebRTC from customer web browser to call center” demonstrates a more elaborate WebRTC architecture, a customer using a web browser to call a corporate call center. When a call is made, the request is sent over the WebSocket connection and the media streams pass through the TURN server. The SIP proxy routes all calls to the corporate PBX which routes the calls to an agent. The media streams must also pass through the PBX for transcoding from the Opus codec to one of the codecs supported by the desk phones, perhaps G.711 or G.722.