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v0.1.0

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@jjaroztegi jjaroztegi released this 18 Oct 11:44
· 61 commits to main since this release
v0.1.0
0ce2064

First public release.

rtpbench calls a voice agent over SIP, captures the RTP on both ends and measures turn latency,
barge-in and SIP behaviour from the packets, on clean and impaired networks (tc netem). Every
timing carries an error bar calibrated against a reference agent, and every case is proven
passing and failing against that agent before it judges yours.

  • Calibration of wire timing, speech detection and barge-in stop time, under sourced network
    profiles, with wall-clock steps logged.
  • 21 conformance cases (greeting, hold, DTMF, transfer, session timers, codec changes, SIP over
    TCP, and four that only a carrier sends).
  • Two bundled targets, LiveKit and AI Voice Connector CE, on a real Realtime engine or a
    deterministic mock, with or without a carrier (OpenSIPS and rtpengine) in front.
  • Live progress for long runs; every run records the image tag of the code it ran.

Published: the baseline,
every case on the reference agent and both targets on the mock engine, direct and through the
carrier, 3 network profiles, 10 repeats, with its raw runs on the
baseline release.

Needs Linux (or WSL2), Docker 28 or later and uv; start with the
quick start.