A WebRTC signaling server is the application’s message channel for exchanging the information two peers need to set up a connection—principally SDP offers and answers, plus ICE candidates. WebRTC does not require a particular signaling server or transport: the application chooses how to carry these messages. ICE, not the signaling server, tests possible network routes and selects a usable one.
What a WebRTC signaling server does
The WebRTC APIs provide mechanisms for establishing peer connections, but they leave signaling to the application. A signaling service routes setup messages between the right participants in a call. It might use a web API, a REST service, RPC, or a messaging channel; “signaling server” names this application role, not one required WebRTC product or protocol. WebRTC.org’s peer-connection guide describes this separation.
The messages include session descriptions and ICE candidates. The server carries them; it does not itself negotiate the network path or inherently carry the call’s audio, video, or data.
How a connection is set up
- The caller creates an offer. The application creates an
RTCPeerConnection, asks it to create an SDP offer, and sets that offer as the local description. - The offer is delivered. The caller sends the offer through the application’s signaling channel to the intended callee.
- The callee creates an answer. The callee sets the received offer as its remote description, creates an SDP answer, sets the answer as its local description, and sends it back over signaling.
- Each peer applies the other description. The caller sets the answer as its remote description. The descriptions let the peers agree on supported session capabilities and the parameters to use; the answer selects a negotiated subset of what the offer describes.
- The peers exchange ICE candidates. Each side sends discovered candidates through signaling, and the other side applies them to its peer connection.
- ICE checks routes. ICE tests candidate pairs and selects a workable route. The application can observe peer-connection state to determine whether a connection has been established.
WebRTC.org’s advanced peer-connections guide covers offer/answer and candidate exchange. SDP is the session description exchanged by the peers; it is not a route-selection service. The W3C WebRTC specification defines the browser-facing peer-connection API.
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Signaling, ICE, STUN, and TURN are different jobs
| Component | Role |
|---|---|
| Signaling channel | Delivers setup messages—such as SDP offers, answers, and ICE candidates—between the applications. |
| ICE | Gathers candidates, exchanges and tests candidate pairs, then selects a usable route. |
| STUN | Can help an ICE agent discover a server-reflexive address, useful when a peer is behind a NAT. |
| TURN | Provides relay candidates and can relay traffic when a direct route is not usable. |
These roles are distinct even if one provider offers both signaling and STUN or TURN services. “Peer-to-peer” does not guarantee every packet travels directly between the two devices: ICE may choose a relayed candidate, in which case TURN relays the traffic. For the protocol details, see IETF RFC 8445, Interactive Connectivity Establishment (ICE).
What Trickle ICE changes
Without trickling, an application may wait for candidate gathering to finish before sending candidates. With Trickle ICE, the ICE agent sends each candidate as it is discovered. That lets connectivity checks begin while gathering continues and can reduce setup delay. The signaling layer must route each incremental candidate to the matching remote peer, which then applies it to the peer connection. WebRTC.org’s guidance on peer connections explains this approach.
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What the application must decide
Because WebRTC does not prescribe a signaling protocol, the application designer must choose a message transport and define how messages map to peers and calls. Useful decision criteria include:
- Delivery and recovery: how the application handles delivery failures, reconnects, and messages arriving after a participant reconnects.
- Routing: how offers, answers, and candidates reach the correct peer and call rather than another session.
- Authentication and authorization: how the application verifies participants and limits who may join or send signaling messages.
- Privacy and retention: what signaling payloads the service stores, for how long, and who can access them.
- Operations: how the service is scaled and kept available for the application’s needs.
- Incremental candidates: whether the design can deliver Trickle ICE candidates as they arrive.
These are design considerations, not a ranking of transports. The WebRTC sources describe the need to exchange negotiation messages but do not declare one signaling technology best for every application.
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