Bridging browsers to SIP is more than translating a signaling protocol. The design has to reconcile browser security, NAT traversal, codec behavior and media encryption with the realities of SIP networks.
Understand the browser security boundary
Browsers require secure signaling and encrypted media. WebSocket transport, DTLS-SRTP and ICE negotiation therefore become first-class parts of the architecture rather than optional features.
Plan NAT traversal deliberately
STUN is useful for discovering reachable addresses, while TURN provides a relay when direct media paths cannot be established. TURN capacity and geographic placement should be treated as media infrastructure, not as a minor utility.
Choose the gateway boundary carefully
The WebRTC-to-SIP boundary may live in a SIP proxy, media server or specialized gateway. The correct placement depends on codec conversion, recording, call control and scaling requirements. Keeping the boundary explicit makes debugging and security policy easier.