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v0.2.0

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@jjaroztegi jjaroztegi released this 22 Oct 20:55
v0.2.0
f3bbeb9

rtpbench now tests an agent of your own by its SIP URI, has a third bundled target, speaks SIP
over TLS and SRTP, and can pass or fail a run in CI.

  • rtpbench target sip:URI runs the cases, latency and barge-in against any agent from the same
    stack, with digest authentication; transfers go to the stack's operator.
  • A third bundled target, Asterisk with AVA, over AudioSocket as upstream ships it or over
    ExternalMedia. AI Voice Connector CE is pinned to upstream main, with its fixes.
  • SIP over TLS and SRTP on the caller's leg and through the carrier, judged from the capture with
    the session keys.
  • Three cases for SIP messages lost on UDP: invite-retransmit, refer-lost and bye-lost.
  • rtpbench check passes or fails a run against a policy, and a GitHub Action runs a target and
    its check on your runner.
  • Reports add how evenly the agent's RTP arrives (split from the path's jitter with
    report --agent-pcap), the audio the mock engine received, per-clip barge-in stop rates with
    the backchannels' words, and where the latency and barge-in turns are.
  • calibrate measures the wall clock's rate and warns above 100 ppm.
  • Every published call's timeline is on the results site.
  • Fixes from a clean-room run of the quick start: --network refuses a Docker network that
    would drop the caller's packets (gateway mode), instead of letting every call time out;
    realtime --port for the mock engine on a host where 8000 is taken; a run stopped with
    SIGTERM (systemd, a cancelled CI job) tears its stack down as Ctrl-C does; one release archive
    can be checked and unpacked without the others.
  • Every number in the published results now has a command that reproduces it from the archives,
    including the baseline's capacity and held-out tables and the AVA page's counts.

Judging changed: clock steps count only where they are large against the error bar; refer is
inconclusive when the transfer itself failed, and timer when the agent does not use session
timers; barge-in leaves out turns whose speech ended before the interjection; RTP is split by the
capturing side's address. Every published set was judged again with these changes, from its
archive.

Published since 0.1.0, each with its raw runs on a release:
metrology,
media path,
SIP over TLS and SRTP,
Asterisk + AVA,
AI Voice Connector CE at b95f5af
and lost SIP messages.

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