Realtime voice latency starts before the model sees a token. It starts when the client opens a live stream, sends small audio frames over an unreliable network, waits for those frames to be buffered, and then waits for generated audio to make the same trip back.
In a typical native integration, your application server becomes the realtime media boundary. The client streams audio to your server, your server buffers it, your server talks to STT, LLM, and TTS providers, and then your server streams audio back to the client. That architecture can work, but the quality of the interaction is now tied to how close your server is to every user.
Allmodels changes the placement of that boundary. The live client stream terminates on a global CDN route close to the user. The chatty part of voice, buffering, jitter, stalls, retries, and backpressure, happens near the client. From there, Allmodels opens cleaner upstream connections to your app and model providers.
This is why a router can be faster than native. The router adds routing overhead, but it can remove a larger client-to-origin latency cost from the realtime loop.
If you already built your own global edge audio router, a native model connection may still be the lowest-latency path. In that case, Allmodels is an abstraction layer: you still get easy model selection, unified billing, fallback, and direct interoperability across Allmodels routes.
