WebCodecs vs WebRTC for recorded calls

Say it plainly first: Synento is slower live than a good WebRTC path. Every frame travels to the relay over a WebSocket from a single region instead of taking the shortest path to each viewer. For a standup, an interview, a support call or a classroom, nobody notices. For rapid-fire conversation across continents, for auctions, for anything where a 300-millisecond delay changes behavior, notice is exactly what they will do.

The honest test is whether your users talk over each other when latency rises. If yes — live debate, negotiation, performance — choose WebRTC and accept the recording machinery that comes with it. If no — and for recorded-first products the answer is usually no — the latency difference buys you an architecture with no media servers, no simulcast tuning, and no separate recording bill.

On a WebRTC platform, recording quality fights the live path. Simulcast sends several quality layers so each viewer gets what their network handles; the recorder must pick layers, handle switches mid-call, composite tiles, and encode the result. Adaptive behavior that helps viewers hurts recordings, and reconciling the two is genuinely hard engineering.

The relay model sidesteps it: one stream per track, the same for every viewer and for storage. What the publisher sent is what the recording holds — camera, microphone and screen on separate timestamped tracks. There is nothing to composite live, so there is nothing to get wrong live. Composition still happens, but at replay and export time, from complete data, where a retry costs milliseconds of compute instead of a lost session.

WebCodecs is newer than WebRTC and reaches fewer browsers: Chrome and Edge 94 and up, Safari 16.4 and up with caveats on iOS, Firefox 133 and up. WebRTC reaches older versions of all of them, plus environments where you do not control the browser at all. If your users join from locked-down corporate machines or old phones, that reach matters more than any recording convenience.

Do not detect this yourself. Gate the live path on the SDK's support check and route everyone else to replay and MP4, which play anywhere. The check below is the whole policy.

Test the matrix the way you test everything else here: run the same two-person call on each browser you claim to support and assert on the manifest afterwards. A browser that joins but never publishes shows up as absent tracks, which turns a vague compatibility complaint into a named browser and a named track.

Choose the WebCodecs relay when the recording is the product: review pipelines, async video, research, support QA. Choose WebRTC when the live moment is the product, or when any of these are hard requirements.

The relay trade is also an operations trade. There are no media servers to size, no SFU clusters to keep balanced across regions, no simulcast policy to tune per network, no recording fleet with its own autoscaling and its own 2am pages. Capacity planning becomes arithmetic on participant-minutes, and the failure surface is one hosted relay instead of a pipeline of stateful machines.

That simplicity is the point of the single-region design, and its limit. One region means predictable behavior and one place to look when something is wrong; it also means a team spread across continents shares whatever latency their distance to the relay implies. If your users cluster in one region — most recorded-first products do — you will never think about this again. If they do not, measure a real call before committing: the manifest plus a stopwatch beats any architecture diagram.