WebRTC vs SIP: Differences and Use Cases

WebRTC and SIP both carry real-time voice, but they were designed for different worlds. SIP grew out of the carrier and enterprise telephony stack, where standardized signaling between vendors and clean interconnect with the PSTN are the reason the protocol exists. WebRTC grew out of the browser, where the goal was letting two web pages send media to each other without anyone installing a plugin. The result is two ecosystems that overlap in capability but disagree on almost every architectural decision underneath.

This guide is a decision-oriented comparison. It covers what each technology actually is, where they differ in practice (signaling, transport, encryption, identity, NAT traversal), the use cases each one wins, and what changes when they have to talk to each other across a boundary. If you want the protocol mechanics of SIP itself, the companion piece SIP Signaling Fundamentals covers the protocol roles and architecture at a higher level, and SIP Call Flow Explained Step by Step walks through the messages on the wire. This article assumes that background and focuses on how WebRTC compares.

One thing this piece does not cover is the full implementation architecture of a WebRTC-to-SIP gateway. That topic deserves its own treatment, and a dedicated gateway architecture article will follow. Here we keep the gateway discussion at the level you need to make decisions, not configure a deployment.

What WebRTC Actually Is

WebRTC is a set of browser APIs (and a matching set of network protocols) that lets a web application capture a microphone or camera, encrypt the stream, and send it to another endpoint with no plugin and no installed client. The reader’s mental model should be: a JavaScript API in the browser, a fixed media stack underneath it, and an explicitly missing piece on top.

The fixed media stack is opinionated. Transport is UDP. Media encryption is SRTP, and it is mandatory; there is no unencrypted mode. Key exchange uses DTLS-SRTP, performed on the media path itself. NAT traversal is baked in through ICE, STUN, and TURN, which together let two browsers behind separate NATs find a working path or fall back to a relay. Codecs are constrained to a small set, with Opus as the audio default and VP8/VP9/H.264/AV1 on the video side.

The missing piece is signaling. WebRTC deliberately leaves out how two endpoints find each other, exchange session descriptions, or learn each other’s ICE candidates. That is left to the application, which usually carries the signaling messages over WebSocket, a custom HTTP scheme, or any other channel the developer chooses. This is the single biggest architectural difference from SIP. SIP is a signaling protocol; WebRTC has none.

The practical consequence is that two browsers running the same web app can talk to each other end to end, but two browsers running different apps cannot. There is no WebRTC equivalent of “dial any SIP URI.” Federation happens at the application layer, not the protocol layer.

The Headline Differences

The following table summarizes the architectural decisions each technology makes. Most of the trade-offs in later sections come back to one of these rows.

Where the Architectures Differ in Practice

The table is a fair summary, but the consequences only become visible once you look at how each design choice plays out in a real deployment.

WebRTC has no signaling protocol

This is the structural difference everything else flows from. A WebRTC application picks its own signaling transport (typically WebSocket carrying JSON), defines its own message formats, and routes messages between users through its own backend. If the application disappears, the signaling disappears with it. SIP is the opposite. The signaling is the standard, and any compliant endpoint can talk to any other compliant endpoint without coordinating in advance with the application that built it.

That property is why SIP is the protocol of interconnect. Two carriers, two PBX vendors, or a PBX and a hosted UCaaS platform can all exchange calls without sharing application code. WebRTC has no equivalent.

SIP decouples signaling from media; WebRTC ties them together by design

In SIP, the signaling path and the media path are independent. SIP negotiates the session (over UDP, TCP, or TLS), and RTP or SRTP flows directly between the endpoints on a different set of ports. The media can take a completely different route through the network from the signaling, and often does.

In WebRTC, the media path is fully specified by the protocol stack: UDP, ICE candidates negotiated through signaling, DTLS handshake on the media socket, SRTP keys derived from that handshake. Signaling is independent in the sense that the application controls it, but the media path is rigidly defined and assumed end to end between the two PeerConnection endpoints.

Mandatory encryption vs optional encryption

WebRTC media is always encrypted. There is no way to negotiate plain RTP between two WebRTC peers. Key exchange happens through DTLS-SRTP on the media socket, which removes the dependency on signaling-layer security for key confidentiality.

SIP allows encrypted media (SRTP, usually keyed by SDES inside an SDP body that travels over TLS-protected signaling) but does not require it. Plenty of carrier and trunk traffic still moves as plain RTP because the operator considers the network trusted. For more on how SRTP key exchange differs between SDES and DTLS-SRTP, see What Is SRTP?.

Identity and trust

SIP has explicit identity machinery. The P-Asserted-Identity header carries an asserted caller identity inside a trusted network, the SIP Identity header (used by STIR/SHAKEN) cryptographically attests to the caller’s number, and carriers maintain trust relationships that give those headers meaning. WebRTC has none of that at the protocol layer. Identity in a WebRTC application is whatever the application chooses to enforce, usually a login token tied to a user account in the same backend that handles signaling.

This matters when WebRTC traffic ends up touching the PSTN. A robocall mitigation framework like STIR/SHAKEN is meaningful inside SIP. It is not meaningful for a browser-to-browser call between two users of the same application.

NAT traversal philosophy

SIP assumes the network operator will solve NAT. The classic answer is to put an SBC at the edge so that internal endpoints register with a public-facing element that handles NAT keepalives, address rewriting, and far-end NAT traversal. SIP ALGs on firewalls try to do something similar in smaller deployments, often with mixed results.

WebRTC assumes the endpoints will solve NAT themselves. ICE walks every candidate address pair (host, server-reflexive via STUN, relayed via TURN), tests them, and picks the best one. A TURN server is the fallback when direct paths fail, and in many large deployments TURN ends up carrying a substantial fraction of media. The benefit is that the protocol works almost anywhere; the cost is that operators have to run STUN and TURN infrastructure.

Use Cases Where SIP Wins

SIP is the protocol you reach for whenever a call needs to leave one organization and arrive at another, whenever it touches the PSTN, or whenever a piece of equipment expects to interconnect with anything other than a copy of itself.

• Carrier interconnect and PSTN access are the original use cases. Every Tier 1 carrier, every ITSP, and every Class 4/5 switch in production today speaks SIP or its SIP-I cousin.
• Microsoft Teams Direct Routing is a SIP integration. Teams Phone uses SIP signaling toward the SBC and SRTP on the media path, with the SBC translating between Teams’ SIP dialect and whatever the carrier delivers. The full mechanics are covered in What Is Teams Direct Routing?.
• Multi-vendor enterprise telephony relies on SIP so that an IP-PBX from one vendor, a contact-center platform from another, and a session border controller from a third can all share trunks.
• IP-PBX deployments and SIP trunking assume SIP from end to end. Even when the desk phones are softphones and the trunk is delivered over the internet, the signaling under the application is SIP.
• Contact-center trunking at scale, including BYOC into Genesys, Five9, or NICE, uses SIP because the carrier side has no other way to deliver inbound calls.

Where there is a number plan, a carrier, or a PBX, SIP is the answer.

Use Cases Where WebRTC Wins

WebRTC’s strength is reaching the user without asking them to install anything. Anywhere the endpoint is a browser, a customer device the operator does not control, or a mobile application that needs a small predictable media stack, WebRTC tends to be the right choice.

• Browser-based softphones for internal employees or remote agents avoid the desktop client problem entirely. A URL and a login are the only deployment surface.
• Click-to-call from a marketing page connects a website visitor to a sales or support queue without dialer software on the visitor’s side.
• In-browser contact-center agent desktops let agents handle calls inside the same CRM tab they live in all day, eliminating a separate softphone client.
• Customer-facing live voice and video support embeds directly into mobile apps and web flows; users do not switch contexts to start a call.
• Internal collaboration apps (the broad category that includes Google Meet, Discord, Zoom’s web client) use WebRTC because the cost of distributing a native client to every participant in a meeting is unacceptable.
• Low-friction onboarding scenarios like telehealth visits, financial advisory consultations, or interview platforms benefit from no-install access for the customer-facing party.

The common pattern is that one side of the conversation is a person on the open internet who must not be asked to install software. WebRTC is the protocol designed for that side.

When They Have to Talk to Each Other

Most real-world deployments are mixed. A browser-based softphone needs to reach the PSTN. A contact-center web client needs to route inbound calls from a SIP trunk. A click-to-call widget needs to drop into a queue served by a traditional ACD. At those points the two worlds have to meet, and four things have to be translated at the boundary.

Signaling is the first translation. The WebRTC side speaks whatever the application chose (commonly WebSocket carrying SIP-over-WebSocket per RFC 7118, or a proprietary JSON protocol). The SIP side speaks standard SIP over UDP, TCP, or TLS. A gateway terminates both and maps between them.

Media encryption is the second. WebRTC requires DTLS-SRTP. The SIP side may deliver SRTP keyed by SDES, or plain RTP. The gateway terminates the DTLS handshake toward the browser and re-keys the media on the other leg using whatever the SIP peer requires.

Codec is the third. Browsers default to Opus; the PSTN and most SIP trunks default to G.711. If both sides support a common codec the call passes through; otherwise the gateway either transcodes or rejects the call. Opus-to-G.711 transcoding is not free; it requires hardware DSP capacity on platforms like ProSBC (the TSBC-HW-TRANS add-on).

NAT traversal is the fourth. The WebRTC side runs ICE against the gateway’s reachable address. The SIP side does not; the gateway terminates ICE on the browser leg and presents a static SIP/RTP endpoint to the carrier or PBX.

A B2BUA-style SBC is the right architectural fit for this boundary because it fully terminates both legs and gives the operator complete control over every header, codec, and crypto context. A SIP proxy cannot do this work; the legs are too different. A dedicated gateway architecture article will cover the implementation details (signaling translation, ICE termination, transcoding placement, scaling).

Where ProSBC Fits

ProSBC is a software B2BUA SBC that terminates the SIP side of deployments where WebRTC traffic arrives upstream. The typical role is the SIP-to-carrier (or SIP-to-PBX) leg: the WebRTC-facing gateway hands a normalized SIP session to ProSBC, and ProSBC handles carrier interop, TLS and SRTP termination toward the trunk, SIP header normalization for the PSTN side, and topology hiding between the cloud and the carrier.

On that leg, ProSBC provides TLS 1.3 for SIP signaling and SRTP (relay or RTP-to-SRTP conversion) for media, with per-trunk-group transport and crypto configuration. The SIP header manipulation engine handles the differences between what a CPaaS or WebRTC gateway produces and what a carrier expects to receive.

ProSBC by itself can perform transcoding for G.711 A-law and mu-law codecs. For more codec support, ProSBC works in tandem with a TSBC-HW-TRANS hardware unit to ensure all systems achieve “real-time” codec negotiation and translation for every single call.

ProSBC does not include a built-in SIP registrar, and it is not the right product for the WebRTC-facing leg itself. The browser-facing gateway (an application server or a WebRTC-aware gateway like Janus or Kamailio with the WebSocket module) sits in front. ProSBC sits behind it on the SIP side.