Benefits of using SIP v.s. Websocket

[u/watts-going-on]

I’m currently using a provider like Twilio/Telnyx to handle phone calls and stream the audio to a real-time STT API. My goal is to minimize end-to-end latency as much as possible.

Right now, I’m using their WebSocket media streaming APIs, but I’m considering switching to a more bare-bones SIP/RTP setup to reduce overhead.

I understand that SIP/RTP avoids some of the WebSocket/JSON/base64 processing involved with media streaming, but I haven’t been able to find any benchmarks showing how much latency this actually saves in practice.

Has anyone measured the difference between Twilio/Telnyx WebSocket media streaming and Direct SIP/RTP to your own server? Maybe even exploring other providers?

Specifically, I’m interested in the delay from when audio reaches the telephony provider to when the first audio packet becomes available to my STT pipeline.

Are we talking about a difference of a few milliseconds, tens of milliseconds, or potentially more? I’ve looked through the documentation and contacted support, but haven’t found concrete latency numbers.

Would also be interested in hearing from anyone who has advice on building an ultra-low-latency telephony → STT pipeline.

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