If you are exploring communication technology for your business or building a product in the telecom space, you have likely encountered two acronyms repeatedly: SIP and WebRTC. While they are often mentioned together, they solve different problems and work in complementary ways.
What is SIP?
The Session Initiation Protocol (SIP) is a signaling protocol used to initiate, maintain, and terminate real-time communication sessions. It has been the backbone of VoIP for over two decades. SIP works with dedicated softphone clients and IP phones, and it integrates seamlessly with existing telephony infrastructure like PBX systems and PSTN gateways.
What is WebRTC?
Web Real-Time Communication (WebRTC) is an open-source project that enables real-time voice, video, and data communication directly in web browsers without plugins. Launched by Google in 2011, it has matured into a powerful standard supported by all major browsers.
Key Differences
Transport
SIP typically uses UDP or TCP for signaling and RTP for media. WebRTC uses DTLS-SRTP for encrypted media transport and ICE/STUN/TURN for NAT traversal. WebRTC is encrypted by default, while SIP encryption (SRTP, TLS) requires explicit configuration.
Infrastructure
SIP requires a SIP server (like Asterisk or FreeSWITCH) and often a registrar and proxy. WebRTC can work peer-to-peer for simple use cases but typically needs a signaling server and TURN relay for production deployments.
Client Requirements
SIP needs a dedicated client application (softphone) or hardware IP phone. WebRTC works natively in any modern browser — no downloads or plugins required.
When to Use Each
Choose SIP when: You need integration with traditional phone networks, your users have dedicated softphone apps, or you require advanced PBX features like call queues and IVR.
Choose WebRTC when: You want browser-based calling with zero setup for end users, you are building click-to-call features on a website, or you need video conferencing embedded in a web application.
The Best of Both Worlds
Many modern platforms combine both. A SIP-based backend handles call routing and PSTN connectivity, while a WebRTC frontend provides easy browser-based access. This hybrid approach gives you the robustness of SIP with the accessibility of WebRTC.
At Incredere Group, our softphone products support both SIP and WebRTC, giving businesses the flexibility to choose what works best for their infrastructure.
