This document discusses best practices for deploying WebRTC to replace or augment existing SIP-based phone systems. It covers choosing appropriate codecs to balance bandwidth usage and call quality for different use cases. It also addresses WebRTC-specific considerations like ICE, DTLS, and asymmetric call patterns. Performance metrics are provided from test calls using different codecs on an Asterisk server. The presentation includes diagrams of common WebRTC deployment architectures and links to live demos.