SIP vs WebRTC: which one does your project actually need?

Teams often frame SIP and WebRTC as an either-or decision. In practice they solve different halves of the same problem. SIP is a signalling protocol for setting up sessions between systems: PBXs, carriers, gateways. WebRTC is a media framework that lets browsers and apps send encrypted audio without plugins. Most modern voice products use both, often on the same call.

Choose SIP when systems talk to systems

• You are connecting a PBX, softswitch or contact centre platform to a carrier.
• You need DIDs, number portability and least-cost routing across providers.
• Your endpoints are desk phones or gateways that already speak SIP.

Choose WebRTC when people join from apps

• Callers or agents work from a browser tab or a mobile app.
• You want encrypted media by default with no client installation.
• You need adaptive codecs like Opus that survive bad Wi-Fi.

The common architecture: both

A typical setup terminates the public phone network over SIP trunks, while agents answer in the browser over WebRTC. A media gateway in the middle transcodes and bridges the two worlds. That is exactly what a modern voice platform does for you, so the practical question is rarely which protocol, but where the bridge should live: in your rack or in the cloud.

Rule of thumb: if you are writing SDP by hand in 2026, you have probably taken on infrastructure that is no longer differentiating. Spend that effort on your call flows instead.