Choose WebRTC when your stream needs real-time interaction and your encoder, ingest service, and playback setup all support it. Choose RTMP or RTMPS when your destination is built around the established encoder-to-ingest workflow and its end-to-end delay is acceptable. Neither protocol guarantees a particular viewer-side latency: the entire publishing, processing, and playback chain determines what people experience.
WebRTC vs RTMP: What is the difference?
WebRTC is a family of technologies for real-time audio and video communication. For publishing into streaming services, WHIP provides a standardized HTTP-based way to ingest WebRTC media. The IETF describes WHIP as a protocol for WebRTC-based ingestion into streaming services or CDNs; the connection process involves ICE, DTLS/SRTP, and RTP/RTCP. Read RFC 9725.
RTMP is a long-established option for sending a live feed from an encoder to an ingest endpoint. RTMPS is the TLS-protected variant in documented workflows. An RTMP ingest choice does not mean viewers necessarily receive RTMP: a service may process the incoming feed and deliver it through a different playback path. AWS, for example, documents both RTMP and WHIP as ways to publish to an IVS stage. AWS IVS ingest options.
| Decision point | WebRTC publishing, commonly via WHIP | RTMP/RTMPS publishing |
|---|---|---|
| Best fit | Interactive, real-time use when the service and playback stack support the required workflow. | Encoder-to-ingest workflows where the destination accepts RTMP/RTMPS and the resulting viewer delay is suitable. |
| Compatibility check | Confirm WHIP support in the encoder and service, plus the service’s media requirements and viewer playback method. | Confirm the destination’s RTMP or RTMPS endpoint, stream-key workflow, and accepted media settings. |
| Network considerations | May need specific TCP and UDP access; firewalls and restricted networks can interfere. | Uses its own destination and port requirements, which also need to be allowed by the network. |
| Latency conclusion | Designed for real-time communication, but no universal end-to-end figure follows from the protocol name alone. | Can be suitable when the full ingest, processing, and playback chain meets the required delay; the protocol name alone does not set viewer latency. |
Which streaming protocol should I use?
Decide against the actual destination and audience experience, not a general claim that one protocol is always better. Check these items before configuring an encoder:
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- Interaction: If guests or viewers must respond in near real time, investigate a WebRTC/WHIP workflow and verify that the playback experience is built for that interaction.
- End-to-end latency target: Establish what delay is acceptable, then check the provider’s ingest, processing, and playback design. There is no neutral, same-conditions comparison in the cited documentation that supports a universal numeric advantage.
- Compatibility: Confirm support across the encoder, ingest provider, and playback client, including any protocol extension or service-specific setup requirement.
- Network access: Ask whether the broadcaster’s firewall permits the required TCP and UDP destinations. A workflow that works on a home network may fail on a corporate or managed network.
- Media constraints: Verify codec, resolution, bitrate, frame rate, keyframe interval, audio/video tracks, and B-frame rules for the particular service.
- Scale and delivery: Choose according to the service’s distribution and playback architecture; protocol labels alone do not describe how viewers receive the stream.
Which protocol has lower latency?
WebRTC is intended for real-time communication and is often the candidate to evaluate when low delay and interaction are central. That does not establish a fixed latency for every WebRTC stream or prove it will outperform every RTMP workflow in practice. Ingest, transcoding or other processing, delivery, player buffering, and network conditions all affect end-to-end delay.
RTMP is a publishing/ingest choice in many encoder workflows. The viewer-facing protocol and buffering behavior may be different from the publisher’s ingest protocol. Compare the complete service-and-player path under your intended configuration rather than treating an ingest protocol as a latency measurement.
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Can I stream with WebRTC from a browser?
Yes, a browser can be part of a WebRTC publishing workflow when the chosen service supplies a compatible setup. Cloudflare documents one browser webcam example that publishes via WHIP and plays via WHEP. That example demonstrates one provider’s workflow, not a guarantee that every browser, destination, or streaming service supports it. See Cloudflare’s browser-based example.
WHIP addresses the publishing side; WHEP is used in Cloudflare’s example for playback. Check the provider’s current browser, media, and network requirements, and test the actual combination of browser, service, and viewers before relying on it for a production stream.
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Does OBS support WebRTC or RTMP?
OBS supports RTMP-style publishing workflows and can publish through WHIP in supported versions and configurations. OBS’s WHIP guide, dated January 14, 2025, explains that a bearer token is the WHIP equivalent of a stream key and that the service details are configured in OBS settings. Because packaging and support can change, check the current OBS release and the service’s own instructions. OBS WHIP Streaming Guide.
OBS with WHIP for Amazon IVS Real-Time
The following are Amazon IVS-specific settings, not general WebRTC or WHIP limits. AWS documents OBS v30 or newer for this workflow, up to 720p and 8.5 Mbps, and recommends a 1- or 2-second keyframe interval. AWS says OBS simulcast is not supported in this workflow and recommends a bitrate below 2.5 Mbps for lower-bandwidth viewers. It also warns that an unstable broadcaster network can cause intermittent freezing. Check the live AWS IVS OBS WHIP guide before setup.
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OBS with RTMP for Amazon IVS
AWS documents a separate RTMP setup for IVS. Its page also gives service-specific 720p and 8.5 Mbps ceilings and keyframe guidance, requires B-frames to be disabled for the described real-time stream, and says the RTMP stream must include audio and video tracks. These are IVS requirements, not protocol-wide requirements. Follow the current AWS IVS RTMP publishing instructions.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.What network access and settings should you check?
Network paths differ by service and protocol. AWS’s IVS network documentation lists TCP 4443 for WebRTC SDP exchange and UDP 32768–61000 for ephemeral WebRTC traffic to its documented destinations; it lists TCP 443 for RTMPS and TCP 1935 for insecure RTMP. These are AWS IVS network requirements, not universal port rules. Recheck the provider’s current documentation and ask the network administrator to allow only the destinations and ports required for your chosen workflow. AWS IVS network requirements.
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Do not copy a bitrate or resolution setting from another platform as if it were a protocol specification. The service may impose its own codec, frame-rate, keyframe, audio, bitrate, resolution, or B-frame requirements. A valid protocol connection can still produce rejected media, poor playback, or instability if those constraints are missed.
How do you choose and verify a workflow?
- Start with the destination. Confirm which ingest protocols and playback modes the service supports, and whether your intended viewer clients can use them.
- Set a viewer-experience target. Decide whether interaction is time-sensitive and what delay is acceptable. Treat this as an end-to-end service requirement, not a promise attached to a protocol label.
- Match the encoder to the ingest. Check the encoder version, endpoint, credentials or stream key/bearer token, and supported media settings in the provider’s documentation.
- Check the network path. Verify that required destinations and TCP/UDP traffic can pass from the broadcaster’s network. Test from the actual venue or office if possible.
- Run a full-path test. Publish a test stream, view it using the actual audience playback method, and assess delay, stability, audio/video presence, and quality under the expected network conditions.
- Keep a fallback plan. If an interactive WebRTC path is unavailable or blocked, determine whether the service’s RTMP/RTMPS ingest is an acceptable alternative, or whether the stream must be rescheduled or moved to a supported network.
Or let it run in the cloud
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Quick Recap
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Product prices and availability are accurate as of the date/time indicated and are subject to change. Any price and availability information displayed on Amazon at the time of purchase will apply.
