A WebRTC signaling server is an application’s message service for passing connection setup information between peers. It carries SDP offers and answers and ICE candidates; WebRTC does not require a particular signaling protocol or server product. ICE—not the signaling server—tests possible network paths and selects one that works.
What a WebRTC signaling server does
WebRTC provides APIs for establishing real-time connections, but leaves signaling to the application. A signaling server is the application-side service that routes the information two peers need to agree on a session. It might use an HTTP-based API, a REST service, RPC, or a messaging channel; “signaling server” names a role, not a standardized protocol or single type of product. WebRTC.org’s peer-connection overview explains this division.
Signaling carries setup messages. It is not inherently the path used to carry audio, video, or data once the connection is established.
How WebRTC signaling works in a call
- The caller creates an offer. The application creates an
RTCPeerConnection, creates an SDP offer describing session capabilities, and sets that offer as the local description. - The offer is sent to the other peer. The application sends the SDP through its signaling channel. The signaling service routes it to the intended callee.
- The callee returns an answer. The callee sets the received offer as its remote description, creates an SDP answer selecting the negotiated subset of capabilities, sets it as the local description, and sends it back through signaling.
- Both peers apply the remote description. Each application supplies the other peer’s SDP description to its peer connection.
- Peers exchange ICE candidates. The applications pass candidates through signaling so each ICE agent can test possible network routes.
- ICE checks connectivity. ICE tests candidate pairs and selects a usable path. The application can monitor the peer connection’s state to determine whether the connection has been established.
SDP describes session capabilities and negotiated parameters; exchanging it does not itself choose the network route. For the browser API’s peer-connection behavior, see the W3C WebRTC specification.
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Signaling, ICE, STUN, and TURN: different jobs
| Component | Role |
|---|---|
| Signaling | Passes setup messages—SDP descriptions and ICE candidates—between the applications. |
| ICE | Gathers candidate addresses, exchanges and tests candidate pairs, and selects a workable route. |
| STUN | Can help an ICE agent discover a server-reflexive address. |
| TURN | Provides relay candidates and can relay traffic when a direct route is not usable. |
These roles are distinct even when one provider offers both signaling and STUN or TURN services. “Peer-to-peer” also does not guarantee that every packet travels directly between devices: ICE can select a relayed route, in which case TURN carries the traffic. The signaling server exchanges setup information; it is not automatically a media server. The IETF’s ICE specification, RFC 8445 (published 2018), describes ICE’s connectivity role.
What trickle ICE changes
Without trickle ICE, an application may wait for candidate gathering to finish before sending the candidate set. With trickle ICE, an ICE agent sends candidates as it discovers them. The remote application applies each arriving candidate to the matching peer connection, allowing connectivity checks to begin while further gathering continues. WebRTC.org describes this incremental exchange as a way to reduce setup delay; it depends on the signaling implementation routing candidates to the correct peer. See WebRTC.org’s advanced peer-connections guide.
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Does WebRTC need a signaling server?
WebRTC needs peers to exchange the information required for negotiation, but the standard does not prescribe a dedicated signaling server or transport. An application must provide some signaling path—for example, an HTTP API or messaging channel—and define how it identifies peers and routes their messages. The service can be custom-built or provided as part of a broader application backend; its exact form depends on the application.
What to consider when building or choosing signaling
Because WebRTC does not standardize the signaling layer, implementation decisions belong to the application. Assess the channel against the demands of your product:
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- Delivery and reconnection: Can peers receive negotiation messages reliably, and what happens if a client disconnects or reconnects mid-call?
- Peer and call routing: How does the service ensure an offer, answer, or candidate reaches the correct peer and session?
- Authentication and authorization: How are users identified, and how is access to a particular call controlled?
- Privacy and retention: What signaling payloads are stored, who can access them, and for how long?
- Scaling and availability: How will the service handle the expected number of concurrent sessions and failures?
- Incremental candidates: Can the channel carry trickled ICE candidates as they arrive and associate them with the correct peer connection?
These are design criteria, not a ranking of particular transports or products: no transport is universally required by WebRTC.
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