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WebRTC supports three distinct live-streaming jobs: connecting two peers for interactive communication, sending a producer’s feed to a media server through WHIP, and delivering a stream from a service to a viewer through WHEP. In every case, signaling exchanges the information peers need to connect; ICE then finds a usable network path. WHIP is a published IETF standard, while the WHEP document cited here is still an Internet-Draft.
How do the three WebRTC workflows differ?
| Workflow | Direction and job | How the session is set up | Status in the cited specification |
|---|---|---|---|
| Peer-to-peer WebRTC | Between two peers for interactive communication | Application-defined signaling exchanges SDP and ICE candidates | WebRTC APIs and the W3C WebRTC Recommendation |
| WHIP | Producer or encoder to a media server, streaming service, or CDN | HTTP offer/answer exchange, followed by ICE, DTLS, and media transmission | IETF RFC 9725, Standards Track, published March 2025 |
| WHEP | Streaming service, CDN, or WebRTC Transmission Network to a viewer | HTTP-based WebRTC session setup | IETF Internet-Draft draft-ietf-wish-whep-04, dated June 22, 2026; not a final RFC |
These are related uses of WebRTC, not interchangeable protocol names. Peer-to-peer describes an interactive connection pattern; WHIP specifies an ingest workflow; WHEP describes an egress workflow. The WHEP revision cited above lists an expiration date of December 24, 2026, and draft text can change before standardization.
How does a peer-to-peer WebRTC connection get established?
The browser or application uses WebRTC peer-connection APIs to negotiate a secure media connection. The WebRTC APIs do not prescribe how the two peers communicate before they are connected, so the application must supply a signaling channel—often an API or RPC connection—to relay setup messages.
1. Exchange an SDP offer and answer
- One peer creates a session description protocol (SDP) offer describing the proposed session and sends it to the other peer using the application’s signaling channel.
- The receiving peer applies the offer, creates an SDP answer, and sends that answer back through signaling.
- Each peer applies the other peer’s description as its remote description. Signaling can relay SDP without interpreting its contents.
An SDP answer means negotiation information has been exchanged; it does not, by itself, prove that a usable media connection exists.
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2. Exchange ICE candidates
Interactive Connectivity Establishment (ICE) gathers possible network paths and tests candidate pairs. As candidates are discovered, each peer sends them over the signaling channel. Apply the remote description before passing remote candidates to the peer connection with addIceCandidate(), as described in MDN’s signaling guidance.
Trickle ICE sends candidates as they become available instead of waiting for gathering to finish. The WebRTC guide says this can reduce connection setup delay. The selected path depends on the participants’ networks and the ICE services available to them.
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3. Configure connectivity services and check connection state
- STUN: commonly participates in the connectivity process.
- TURN: provides a relay path when a direct path is not available.
- Connection state: monitor the peer connection and confirm that a candidate pair has been nominated and the secure media connection established before treating the session as connected.
STUN and TURN server configuration is supplied to the peer connection. The W3C WebRTC Recommendation and the WebRTC guide describe ICE as the mechanism that works with these services to handle differing network conditions; the exact route is not guaranteed in advance.
How do I send a WebRTC stream to a server with WHIP?
WHIP—the WebRTC-HTTP Ingestion Protocol—standardizes an HTTP-based way for a producer, such as an encoder or media producer, to send a WebRTC feed into streaming infrastructure. IETF RFC 9725, published as a Standards Track document in March 2025, defines the protocol for ingest into streaming services and/or CDNs.
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- Obtain the WHIP endpoint. The streaming service or media server provides the endpoint the client will use. A WHIP endpoint may also provide STUN/TURN configuration in its successful response.
- POST an SDP offer. The client sends its offer to the endpoint over HTTP. The endpoint returns a successful response containing an SDP answer.
- Complete connectivity and security setup. The client and endpoint establish ICE and DTLS. The client then sends media to the server as RTP/RTCP protected with SRTP.
- Handle ICE updates when needed. WHIP’s HTTP PATCH mechanism can carry ICE-related updates, including trickle ICE or ICE restarts. RFC 9725 recommends that WHIP sessions and clients support trickle ICE and ICE restarts.
WHIP deliberately limits renegotiation: after the initial negotiation, it does not support SDP renegotiation or changes to media sections. It is therefore important to distinguish an ICE update from a request to renegotiate the session’s media description.
How does a viewer play a WebRTC live stream with WHEP?
WHEP—the WebRTC-HTTP Egress Protocol—is intended for the viewer-facing direction: a streaming service, CDN, or WebRTC Transmission Network supplies a live stream to a viewer using HTTP-based WebRTC setup. The cited document, draft-ietf-wish-whep-04, describes a one-time SDP offer/answer exchange and ICE-related PATCH updates, with no SDP renegotiation after the initial exchange.
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Status matters: the cited WHEP document is an IETF Internet-Draft, revision -04 dated June 22, 2026, not a finalized RFC. Its listed expiration date is December 24, 2026. Treat its specified behavior as draft protocol text, not as a guarantee that every service implements it or that the text will remain unchanged.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.What commonly goes wrong during setup?
The following checks follow the negotiation and connectivity sequence described by the WebRTC guide, MDN signaling guidance, the W3C WebRTC Recommendation, and RFC 9725. They are diagnostic starting points, not a claim that every failure has a single cause.
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- The SDP answer arrives, but media does not connect: an answer is not proof of a completed connection. Check the peer-connection state and whether ICE has nominated a candidate pair.
- Remote candidate handling fails or connectivity stalls: verify that the remote description has been applied before calling
addIceCandidate(), and confirm the application is relaying candidates to the intended peer. - No direct network path succeeds: review the ICE server configuration and whether TURN relay connectivity is available. Network conditions can prevent a direct path from working.
- Setup waits for candidate gathering to finish: consider trickle ICE so candidates can be exchanged as they are discovered. It can reduce setup delay, but does not guarantee a connection.
- A WHIP session needs an ICE change: use the protocol’s ICE-related PATCH mechanism where supported; do not treat it as SDP renegotiation, which WHIP does not support after initial negotiation.
- The stream works for one network but not another: recheck ICE services and candidate-pair selection under the affected network conditions. The path that works in one environment may not work in another.
What WebRTC does not decide for a live-streaming project
These workflows explain session setup and media transport; they do not by themselves select an encoder, define a particular platform’s deployment, or establish copyright permission. For material you broadcast, make sure you have the rights and permissions needed. A protocol carrying a recording does not grant those rights, and technical delivery does not determine whether content meets a platform’s monetization or reused-content requirements.
The cited specifications also do not establish latency benchmarks or prescribe how a service should handle recording, transcoding, or large-scale forwarding. Those are separate implementation and service-design questions; do not infer a performance result from the protocol workflow alone.
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