rtpbench 0.3.0__tar.gz

rtpbench-0.3.0/.dockerignore

@@ -0,0 +1,4 @@
+*
+!src/rtpbench/*.py
+!*.py
+!corpus/**/*.wav

rtpbench-0.3.0/.gitignore

@@ -0,0 +1,14 @@
+__pycache__/
+.venv*/
+.*_cache/
+dist/
+*.pcap
+*.pcapng
+results/*
+!results/baseline/
+!results/metrology/
+!results/media-path/
+!results/tls-srtp/
+!results/ava/
+!results/aivc-b95f5af/
+!results/retransmit/

rtpbench-0.3.0/Dockerfile

@@ -0,0 +1,12 @@
+FROM python:3.14-slim@sha256:a2b82f3c48559aa0a8446d9af49826b6e2b2016f4cd2afabfe6013ec53729170
+RUN apt-get update \
+    && apt-get install -y --no-install-recommends iproute2 iptables tcpdump \
+    && rm -rf /var/lib/apt/lists/*
+# TLS records and SRTP need AES, which the standard library lacks; pinned as uv.lock pins them
+RUN pip install --no-cache-dir cryptography==50.0.2 cffi==2.1.1 pycparser==3.11
+# the package in the build context: a checkout's src/rtpbench, or an installed rtpbench's own
+# directory (.), which holds this file and the corpus too
+ARG RTPBENCH_SRC=src/rtpbench
+COPY ${RTPBENCH_SRC}/*.py /app/src/rtpbench/
+COPY corpus /app/corpus
+ENV PYTHONPATH=/app/src PYTHONUNBUFFERED=1

rtpbench-0.3.0/LICENSE

@@ -0,0 +1,202 @@
+
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rtpbench-0.3.0/PKG-INFO

@@ -0,0 +1,123 @@
+Metadata-Version: 2.5
+Name: rtpbench
+Version: 0.3.0
+Summary: Wire-level latency and conformance tests for voice agents over SIP/RTP
+Author-email: Juanjo Aroztegi <juanjo@aroztegi.com>
+License-Expression: Apache-2.0
+License-File: LICENSE
+Requires-Python: >=3.14
+Requires-Dist: cryptography>=50
+Requires-Dist: rich>=15
+Provides-Extra: vad
+Requires-Dist: onnxruntime>=1.31; extra == 'vad'
+Requires-Dist: webrtcvad-wheels>=2.0.14; extra == 'vad'
+Description-Content-Type: text/markdown
+
+# rtpbench
+
+Wire-level latency and conformance tests for voice agents over SIP/RTP.
+
+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 themselves, on clean and impaired networks
+(`tc netem`). Every measurement carries an error bar calibrated against a reference agent that
+logs the truth, and every case is proven against that agent, passing and failing:
+`rtpbench cases` runs that proof on your host.
+
+Status: early; commands and result formats can still change between releases.
+
+## What it found
+
+Three open-source ways to put the OpenAI Realtime API on a SIP call, through 24 cases, with and
+without a carrier in front, on a mock engine that holds the AI engine's share of each turn fixed:
+
+| | [LiveKit](https://github.com/livekit/sip) | [AI Voice Connector CE](https://github.com/OpenSIPS/opensips-ai-voice-connector-ce) at b95f5af | Asterisk + [AVA](https://github.com/hkjarral/AVA-AI-Voice-Agent-for-Asterisk) |
+| --- | --- | --- | --- |
+| cases failed of 24 | 5, and 2 unsupported | 5, 3 unsupported and 1 inconclusive | 4, and 3 inconclusive |
+| turn latency, clean, p50 | 1833 ms | 1511 ms | 1588 ms |
+| its own media path per turn, p50 | 335 ms | 9 ms | 88 ms |
+| stops for an interruption, clean, p50 | 306 ms | 24 ms | never |
+
+The counts take every profile, direct and through the carrier. Run direct only, as
+`target NAME --mock` does by default, LiveKit fails 4: `late-offer` fails only through the
+carrier. All three pass the 3 cases for lost SIP messages, run direct on a clean network.
+Unsupported is an optional feature declined as its RFC allows, with the call unharmed:
+LiveKit refuses DTMF by SIP INFO (405), AI Voice Connector CE refuses it (501) and negotiates no
+RFC 4733 events, and neither takes session timers. LiveKit does negotiate RFC 4733 events, but
+the digit never reaches the engine: a failure.
+
+- LiveKit keeps sending audio to a held caller in every kind of hold, AI Voice Connector CE when
+  held with `c=0.0.0.0`, and both answer a re-INVITE that changes codec with the old codec. AVA
+  sends RTP only while it speaks, so whether its media resumes after a hold cannot be judged.
+- LiveKit's extra latency is its own media path: 326 ms per turn more than AI Voice Connector
+  CE's, against 322 ms between their turn latencies.
+- LiveKit's RTP falls 20 ms behind its own timestamps 4 s into every call, so a receiver needs a
+  21 ms jitter buffer for it on a clean network; AI Voice Connector CE's needs 1.1 ms.
+- AI Voice Connector CE delivers the engine's speech sample for sample. LiveKit re-encodes it on
+  the way, through its WebRTC room, and on a clean network stretches or cuts a pause by 20-40 ms
+  in 6% of answers (10% in a second wave).
+- AVA, as its golden configuration ships, cuts the last 0.8 s of every answer, and it never
+  stops for an interruption: it keeps the caller's audio from the engine while it speaks. Over
+  AudioSocket under packet loss, the engine sometimes got silence instead of the caller, so 5 more
+  cases fail now and then.
+- AI Voice Connector CE's main branch fixed 4 of the 10 cases it failed in the baseline (a failed
+  transfer no longer hangs up on the caller; compact headers, a bare `a=ssrc` and SIP over TCP
+  are taken), and now declines a DTMF INFO with 501 where it left it unanswered. It still
+  refuses an INVITE without SDP and answers no re-INVITE without one.
+
+The details, commands, conditions and raw runs are in [the baseline](results/baseline),
+[AI Voice Connector CE at b95f5af](results/aivc-b95f5af), [Asterisk + AVA](results/ava) and
+[lost SIP messages](results/retransmit), and every call's timeline on
+[the results site](https://jjaroztegi.github.io/rtpbench/).
+
+Common ways of measuring turn latency have errors of their own: measured inside the agent, a turn
+misses the network (80 ms on an LTE path); an energy rule on a recording reads turns long, by
+87-110 ms at the median. Each method against the truth: [metrology](results/metrology).
+
+## Quick start
+
+Needs Linux, Docker 28 or later with Compose, and [uv](https://docs.astral.sh/uv/). WSL2 runs
+it, but its clock can drift and step by far more than the error bars: read what `calibrate`
+warns about first ([clock steps and rate](docs/calibration.md#clock-steps-and-rate)).
+
+```sh
+uvx rtpbench calibrate --profiles clean --turns 5     # how exactly this host times the wire
+uvx 'rtpbench[vad]' cases                             # prove every case against the reference agent
+uvx 'rtpbench[vad]' target livekit --mock             # a bundled target, no AI account needed
+uvx 'rtpbench[vad]' target sip:agent@host             # an agent of your own, the same way
+```
+
+The first three take about 1, 17 and 11 minutes, after pulling about 0.4 GB of Docker images
+(compressed) on first use; every target and the carrier take 1.5 GB
+([images](docs/README.md#images)). The last needs an agent that answers SIP; [your own
+agent](docs/targets.md#your-own-agent) says what it needs, such as `--network` for one in Docker on
+this host.
+
+Each run writes `results/<timestamp>/` with the captures, `summary.md` and, for cases,
+`report.html`: a timeline of every call's SIP, speech and RTP. `rtpbench index` links every run
+under `results/` from `results/index.html`; `rtpbench COMMAND --help` lists a command's options,
+defaults and valid names.
+
+## Documentation
+
+- [Calibration](docs/calibration.md): error bars, clock steps, network profiles and their sources
+- [Conformance cases](docs/cases.md): what each case checks
+- [Targets, your own agent, the mock engine and the carrier](docs/targets.md)
+- [CI](docs/ci.md): pass or fail your agent's run against a policy
+
+## Development
+
+From a clone, `uv run rtpbench` (`uv run --extra vad rtpbench` for `[vad]`) runs the checkout's
+code, as the docs show it:
+
+```sh
+uv sync
+uv run ruff format && uv run ruff check --fix
+uv run mypy
+uv run pytest
+```
+
+## Licence
+
+Apache-2.0, except `targets/aivc/opensips.cfg`, derived from AI Voice Connector CE's example and
+under its GPL-3.0-or-later, and `targets/ava/ai-agent.yaml`, derived from AVA's golden OpenAI
+configuration and under its MIT licence.

rtpbench-0.3.0/README.md

@@ -0,0 +1,108 @@
+# rtpbench
+
+Wire-level latency and conformance tests for voice agents over SIP/RTP.
+
+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 themselves, on clean and impaired networks
+(`tc netem`). Every measurement carries an error bar calibrated against a reference agent that
+logs the truth, and every case is proven against that agent, passing and failing:
+`rtpbench cases` runs that proof on your host.
+
+Status: early; commands and result formats can still change between releases.
+
+## What it found
+
+Three open-source ways to put the OpenAI Realtime API on a SIP call, through 24 cases, with and
+without a carrier in front, on a mock engine that holds the AI engine's share of each turn fixed:
+
+| | [LiveKit](https://github.com/livekit/sip) | [AI Voice Connector CE](https://github.com/OpenSIPS/opensips-ai-voice-connector-ce) at b95f5af | Asterisk + [AVA](https://github.com/hkjarral/AVA-AI-Voice-Agent-for-Asterisk) |
+| --- | --- | --- | --- |
+| cases failed of 24 | 5, and 2 unsupported | 5, 3 unsupported and 1 inconclusive | 4, and 3 inconclusive |
+| turn latency, clean, p50 | 1833 ms | 1511 ms | 1588 ms |
+| its own media path per turn, p50 | 335 ms | 9 ms | 88 ms |
+| stops for an interruption, clean, p50 | 306 ms | 24 ms | never |
+
+The counts take every profile, direct and through the carrier. Run direct only, as
+`target NAME --mock` does by default, LiveKit fails 4: `late-offer` fails only through the
+carrier. All three pass the 3 cases for lost SIP messages, run direct on a clean network.
+Unsupported is an optional feature declined as its RFC allows, with the call unharmed:
+LiveKit refuses DTMF by SIP INFO (405), AI Voice Connector CE refuses it (501) and negotiates no
+RFC 4733 events, and neither takes session timers. LiveKit does negotiate RFC 4733 events, but
+the digit never reaches the engine: a failure.
+
+- LiveKit keeps sending audio to a held caller in every kind of hold, AI Voice Connector CE when
+  held with `c=0.0.0.0`, and both answer a re-INVITE that changes codec with the old codec. AVA
+  sends RTP only while it speaks, so whether its media resumes after a hold cannot be judged.
+- LiveKit's extra latency is its own media path: 326 ms per turn more than AI Voice Connector
+  CE's, against 322 ms between their turn latencies.
+- LiveKit's RTP falls 20 ms behind its own timestamps 4 s into every call, so a receiver needs a
+  21 ms jitter buffer for it on a clean network; AI Voice Connector CE's needs 1.1 ms.
+- AI Voice Connector CE delivers the engine's speech sample for sample. LiveKit re-encodes it on
+  the way, through its WebRTC room, and on a clean network stretches or cuts a pause by 20-40 ms
+  in 6% of answers (10% in a second wave).
+- AVA, as its golden configuration ships, cuts the last 0.8 s of every answer, and it never
+  stops for an interruption: it keeps the caller's audio from the engine while it speaks. Over
+  AudioSocket under packet loss, the engine sometimes got silence instead of the caller, so 5 more
+  cases fail now and then.
+- AI Voice Connector CE's main branch fixed 4 of the 10 cases it failed in the baseline (a failed
+  transfer no longer hangs up on the caller; compact headers, a bare `a=ssrc` and SIP over TCP
+  are taken), and now declines a DTMF INFO with 501 where it left it unanswered. It still
+  refuses an INVITE without SDP and answers no re-INVITE without one.
+
+The details, commands, conditions and raw runs are in [the baseline](results/baseline),
+[AI Voice Connector CE at b95f5af](results/aivc-b95f5af), [Asterisk + AVA](results/ava) and
+[lost SIP messages](results/retransmit), and every call's timeline on
+[the results site](https://jjaroztegi.github.io/rtpbench/).
+
+Common ways of measuring turn latency have errors of their own: measured inside the agent, a turn
+misses the network (80 ms on an LTE path); an energy rule on a recording reads turns long, by
+87-110 ms at the median. Each method against the truth: [metrology](results/metrology).
+
+## Quick start
+
+Needs Linux, Docker 28 or later with Compose, and [uv](https://docs.astral.sh/uv/). WSL2 runs
+it, but its clock can drift and step by far more than the error bars: read what `calibrate`
+warns about first ([clock steps and rate](docs/calibration.md#clock-steps-and-rate)).
+
+```sh
+uvx rtpbench calibrate --profiles clean --turns 5     # how exactly this host times the wire
+uvx 'rtpbench[vad]' cases                             # prove every case against the reference agent
+uvx 'rtpbench[vad]' target livekit --mock             # a bundled target, no AI account needed
+uvx 'rtpbench[vad]' target sip:agent@host             # an agent of your own, the same way
+```
+
+The first three take about 1, 17 and 11 minutes, after pulling about 0.4 GB of Docker images
+(compressed) on first use; every target and the carrier take 1.5 GB
+([images](docs/README.md#images)). The last needs an agent that answers SIP; [your own
+agent](docs/targets.md#your-own-agent) says what it needs, such as `--network` for one in Docker on
+this host.
+
+Each run writes `results/<timestamp>/` with the captures, `summary.md` and, for cases,
+`report.html`: a timeline of every call's SIP, speech and RTP. `rtpbench index` links every run
+under `results/` from `results/index.html`; `rtpbench COMMAND --help` lists a command's options,
+defaults and valid names.
+
+## Documentation
+
+- [Calibration](docs/calibration.md): error bars, clock steps, network profiles and their sources
+- [Conformance cases](docs/cases.md): what each case checks
+- [Targets, your own agent, the mock engine and the carrier](docs/targets.md)
+- [CI](docs/ci.md): pass or fail your agent's run against a policy
+
+## Development
+
+From a clone, `uv run rtpbench` (`uv run --extra vad rtpbench` for `[vad]`) runs the checkout's
+code, as the docs show it:
+
+```sh
+uv sync
+uv run ruff format && uv run ruff check --fix
+uv run mypy
+uv run pytest
+```
+
+## Licence
+
+Apache-2.0, except `targets/aivc/opensips.cfg`, derived from AI Voice Connector CE's example and
+under its GPL-3.0-or-later, and `targets/ava/ai-agent.yaml`, derived from AVA's golden OpenAI
+configuration and under its MIT licence.

rtpbench-0.3.0/carrier/compose.yaml

@@ -0,0 +1,68 @@
+# A carrier between the router and the agent (the reference agent or a target), merged over
+# ../compose.yaml: OpenSIPS as an SBC (stateful proxy with topology hiding) and rtpengine
+# anchoring the media, as real trunks put in front of a PBX. `carrier` owns the network namespace
+# on agentnet, so caller <-> carrier crosses the netem router and carrier <-> agent does not.
+# rtpbench sets RTPBENCH_CARRIER_NEXT to the agent's host (and RTPBENCH_CARRIER_PORT, its SIP
+# port, for an agent outside the stack, which the carrier reaches through Docker's gateway).
+# Paths are relative to the first compose file's directory.
+services:
+  carrier:
+    image: ${RTPBENCH_IMAGE:?}
+    cap_add: [NET_ADMIN]
+    init: true
+    depends_on: [router]
+    networks: [agentnet]
+    volumes: ["${RTPBENCH_OUT:?}:/out"]
+    command: sh -c 'ip route replace default via $$(getent hosts router | cut -d" " -f1) && exec sleep infinity'
+
+  carrier-media:
+    build:
+      # neither build copies a file, but the tag hashes the context's: not the whole repository
+      context: ./carrier
+      dockerfile_inline: |
+        FROM debian:trixie@sha256:913f6706df59a68922d1dd08f78c2476560a8d367897200a6005b00e5f67c2d5
+        RUN apt-get update \
+            && apt-get install -y --no-install-recommends rtpengine-daemon=12.5.1.31-1 \
+            && rm -rf /var/lib/apt/lists/*
+    image: ${RTPBENCH_REGISTRY:?}/rtpbench-rtpengine:12.5.1.31${RTPBENCH_BUILD_CARRIER_MEDIA:-}
+    network_mode: service:carrier
+    depends_on: [carrier]
+    # userspace forwarding (no kernel module in a container), media on the agentnet address
+    command: >-
+      rtpengine --foreground --log-stderr --table=-1 --interface=eth0
+      --listen-ng=127.0.0.1:2223 --port-min=30000 --port-max=30999
+    healthcheck:  # its control socket (127.0.0.1:2223) bound; OpenSIPS disables a relay that
+      # does not answer its first ping, for a minute, and calls then bypass it
+      test: grep -q ' 0100007F:08AF 00000000:0000 07 ' /proc/net/udp
+      interval: 1s
+      retries: 20
+
+  carrier-sip:
+    build:  # the same OpenSIPS, with its digest authentication and TLS modules (not in the image)
+      context: ./carrier
+      dockerfile_inline: |
+        FROM opensips/opensips:3.6@sha256:81a7fa3c7bdbceaadb7cb869ff7ac3f34efb2d95cd7fecb3b689d7b6b1eb0672
+        RUN apt-get update \
+            && apt-get install -y --no-install-recommends opensips-auth-modules=3.6.9-1 \
+                opensips-tls-module=3.6.9-1 opensips-tls-openssl-module=3.6.9-1 \
+                opensips-tlsmgm-module=3.6.9-1 \
+            && rm -rf /var/lib/apt/lists/*
+    image: ${RTPBENCH_REGISTRY:?}/rtpbench-opensips:3.6.9-1-tls${RTPBENCH_BUILD_CARRIER_SIP:-}
+    network_mode: service:carrier
+    depends_on:
+      carrier-media: {condition: service_healthy}
+    volumes: ["./carrier/opensips.cfg:/etc/opensips/opensips.cfg:ro"]
+    environment:
+      CARRIER_NEXT: ${RTPBENCH_CARRIER_NEXT:?}
+      CARRIER_NEXT_PORT: ${RTPBENCH_CARRIER_PORT:-5060}
+      CARRIER_NEXT_TRANSPORT: ${RTPBENCH_CARRIER_TRANSPORT:-udp}
+      # set, the carrier demands these credentials of the caller, as a trunk may
+      CARRIER_AUTH_USER: ${RTPBENCH_CARRIER_AUTH_USER:-}
+      CARRIER_AUTH_PASSWORD: ${RTPBENCH_CARRIER_AUTH_PASSWORD:-}
+      # set, it answers the agent's challenge with these
+      CARRIER_UAC_USER: ${RTPBENCH_CARRIER_UAC_USER:-}
+      CARRIER_UAC_PASSWORD: ${RTPBENCH_CARRIER_UAC_PASSWORD:-}
+    healthcheck:
+      test: opensips-cli -x mi sr_get_status core | grep -q running
+      interval: 1s
+      retries: 20