PyPI - sipstuff - Versions diffs - 0.0.2__tar.gz → 0.0.4__tar.gz - Mend

sipstuff 0.0.2tar.gz → 0.0.4tar.gz

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{sipstuff-0.0.2 → sipstuff-0.0.4}/PKG-INFO RENAMED Viewed

@@ -1,6 +1,6 @@
-Metadata-Version: 2.4
+Metadata-Version: 2.5
 Name: sipstuff
-Version: 0.0.2
+Version: 0.0.4
 Summary: SIP telephony automation toolkit: place calls via PJSIP, play WAV/TTS audio, record and transcribe responses
 Project-URL: Homepage, https://github.com/vroomfondel/sipstuff
 Project-URL: Repository, https://github.com/vroomfondel/sipstuff
@@ -18,15 +18,15 @@ Classifier: Topic :: Communications :: Telephony
 Classifier: Topic :: Multimedia :: Sound/Audio :: Speech
 Requires-Python: >=3.14
 Requires-Dist: faster-whisper
-Requires-Dist: loguru>=0.7.0
+Requires-Dist: loguru>=0.7.3
 Requires-Dist: numpy
-Requires-Dist: piper-tts>=1.4.0
-Requires-Dist: pydantic>=2.10.0
-Requires-Dist: pyyaml>=6.0
+Requires-Dist: piper-tts>=1.8.0
+Requires-Dist: pydantic>=2.14.2
+Requires-Dist: pyyaml>=6.0.3
 Requires-Dist: ruamel-yaml
 Requires-Dist: sounddevice
 Requires-Dist: soundfile
-Requires-Dist: tabulate>=0.9.0
+Requires-Dist: tabulate>=0.10.0
 Provides-Extra: cuda
 Requires-Dist: nvidia-cublas-cu12; extra == 'cuda'
 Requires-Dist: nvidia-cudnn-cu12; extra == 'cuda'
@@ -42,6 +42,7 @@ Description-Content-Type: text/markdown
 [![black-lint](https://github.com/vroomfondel/sipstuff/actions/workflows/checkblack.yml/badge.svg)](https://github.com/vroomfondel/sipstuff/actions/workflows/checkblack.yml)
 ![Cumulative Clones](https://img.shields.io/endpoint?logo=github&url=https://gist.githubusercontent.com/vroomfondel/92ea75186bc004c1125824335f69a821/raw/sipstuff_clone_count.json)
 [![Docker Pulls](https://img.shields.io/docker/pulls/xomoxcc/sipstuff?logo=docker)](https://hub.docker.com/r/xomoxcc/sipstuff/tags)
+[![PyPI Downloads](https://static.pepy.tech/personalized-badge/sipstuff?period=total&units=INTERNATIONAL_SYSTEM&left_color=BLACK&right_color=GREEN&left_text=PyPi+Downloads)](https://pepy.tech/projects/sipstuff)
 [![PyPI](https://img.shields.io/pypi/v/sipstuff?logo=pypi&logoColor=white)](https://pypi.org/project/sipstuff/)
 [![Gemini_Generated_Image_cpliijcpliijcpli_250x250.png](https://raw.githubusercontent.com/vroomfondel/sipstuff/main/Gemini_Generated_Image_cpliijcpliijcpli_250x250.png)](https://hub.docker.com/r/xomoxcc/sipstuff/tags)
@@ -179,18 +180,18 @@ The `--json` flag outputs structured JSON including segment timestamps, audio du
 }
 ```
-| Flag | Description |
-|------|-------------|
-| `wav` (positional) | Path to WAV file to transcribe |
-| `--backend` | STT backend: `faster-whisper` (default) or `openvino` |
-| `--model`, `-m` | Whisper model size or HuggingFace model ID (default: `medium`) |
-| `--language`, `-l` | Language code (default: `de`) |
-| `--device` | Compute device: `cpu` or `cuda` (default: `cpu`) |
-| `--compute-type` | Quantization: `int8`, `float16`, `float32` (auto-selected by default) |
-| `--data-dir` | Directory for Whisper models (default: `~/.local/share/faster-whisper-models`) |
-| `--json` | Output result as JSON with metadata and segment timestamps |
-| `--no-vad` | Disable Silero VAD pre-filtering (VAD is on by default, recommended for phone recordings) |
-| `--verbose`, `-v` | Debug logging |
+| Flag               | Description                                                                               |
+|--------------------|-------------------------------------------------------------------------------------------|
+| `wav` (positional) | Path to WAV file to transcribe                                                            |
+| `--backend`        | STT backend: `faster-whisper` (default) or `openvino`                                     |
+| `--model`, `-m`    | Whisper model size or HuggingFace model ID (default: `medium`)                            |
+| `--language`, `-l` | Language code (default: `de`)                                                             |
+| `--device`         | Compute device: `cpu` or `cuda` (default: `cpu`)                                          |
+| `--compute-type`   | Quantization: `int8`, `float16`, `float32` (auto-selected by default)                     |
+| `--data-dir`       | Directory for Whisper models (default: `~/.local/share/faster-whisper-models`)            |
+| `--json`           | Output result as JSON with metadata and segment timestamps                                |
+| `--no-vad`         | Disable Silero VAD pre-filtering (VAD is on by default, recommended for phone recordings) |
+| `--verbose`, `-v`  | Debug logging                                                                             |
 ### `call` — Place a SIP Call
@@ -212,7 +213,7 @@ python -m sipstuff.cli call \
     --server pbx.local --user 1000 --password secret \
     --dest +491234567890 \
     --text "Achtung! Wasserstand kritisch!" \
-    --tts-model de_DE-thorsten-high --tts-sample-rate 8000
+    --piper-model de_DE-thorsten-high --tts-sample-rate 8000
 # Wait for callee to finish speaking before playback
 python -m sipstuff.cli call \
@@ -271,7 +272,7 @@ python -m sipstuff.cli call \
     --server 192.168.1.100 --user 1001 --password secret \
     --dest +491234567890 \
     --interactive \
-    --piper-model /path/to/de_DE-thorsten-high.onnx \
+    --piper-live-model de_DE-thorsten-high \
     --text "Hallo, hier spricht die Maschine." \
     --play-audio --play-tx --real-capture -v
@@ -295,102 +296,105 @@ When `--transcribe` is used with `--record`, a JSON call report is written next
 **SIP connection:**
-| Flag | Description |
-|------|-------------|
-| `--config`, `-c` | Path to YAML config file |
-| `--server`, `-s` | PBX hostname or IP |
-| `--port`, `-p` | SIP port (default: 5060) |
-| `--user`, `-u` | SIP extension / username |
-| `--password` | SIP password |
-| `--transport` | `udp`, `tcp`, or `tls` (default: udp) |
-| `--srtp` | `disabled`, `optional`, or `mandatory` (default: disabled) |
-| `--tls-verify` | Verify TLS server certificate |
-| `--dest`, `-d` | Destination phone number or SIP URI (required) |
-| `--timeout`, `-t` | Call timeout in seconds (default: 60) |
+| Flag                 | Description                                                |
+|----------------------|------------------------------------------------------------|
+| `--config`, `-c`     | Path to YAML config file                                   |
+| `--server`, `-s`     | PBX hostname or IP                                         |
+| `--port`, `-p`       | SIP port (default: 5060)                                   |
+| `--user`, `-u`       | SIP extension / username                                   |
+| `--password`         | SIP password                                               |
+| `--transport`        | `udp`, `tcp`, or `tls` (default: udp)                      |
+| `--srtp`             | `disabled`, `optional`, or `mandatory` (default: disabled) |
+| `--tls-verify`       | Verify TLS server certificate                              |
+| `--rtp-port`         | RTP start port, 0 = OS-assigned (default: 4000)            |
+| `--rtp-port-range`   | RTP port window width (default: 200)                       |
+| `--no-rtp-randomize` | Always start at `--rtp-port` instead of a random offset    |
+| `--dest`, `-d`       | Destination phone number or SIP URI (required)             |
+| `--timeout`, `-t`    | Call timeout in seconds (default: 60)                      |
 **Audio source (mutually exclusive):**
-| Flag | Description |
-|------|-------------|
-| `--wav`, `-w` | Path to WAV file to play (mutually exclusive with `--interactive`) |
-| `--interactive` | Interactive live TTS mode: type text in the console that gets spoken via Piper TTS during the call (mutually exclusive with `--wav`, requires `--piper-model`) |
+| Flag            | Description                                                                                                                                                         |
+|-----------------|---------------------------------------------------------------------------------------------------------------------------------------------------------------------|
+| `--wav`, `-w`   | Path to WAV file to play (mutually exclusive with `--interactive`)                                                                                                  |
+| `--interactive` | Interactive live TTS mode: type text in the console that gets spoken via Piper TTS during the call (mutually exclusive with `--wav`, requires `--piper-live-model`) |
 **TTS:**
-| Flag | Description |
-|------|-------------|
-| `--text` | Text to synthesize via piper TTS, or initial greeting in interactive mode |
-| `--tts-model` | Piper voice model for pre-generated TTS (default: `de_DE-thorsten-high`) |
-| `--piper-model` | Path to Piper `.onnx` model for live TTS in interactive mode |
-| `--tts-sample-rate` | Resample TTS output to this rate in Hz (default: native/22050) |
-| `--tts-data-dir` | Directory for piper voice models (default: `~/.local/share/piper-voices`) |
-| `--tts-cuda` | Use CUDA GPU acceleration for Piper TTS |
+| Flag                 | Description                                                               |
+|----------------------|---------------------------------------------------------------------------|
+| `--text`             | Text to synthesize via piper TTS, or initial greeting in interactive mode |
+| `--piper-model`      | Piper voice model for pre-generated TTS (default: `de_DE-thorsten-high`)  |
+| `--piper-live-model` | Piper voice model for live TTS in interactive mode                        |
+| `--tts-sample-rate`  | Resample TTS output to this rate in Hz (default: native/22050)            |
+| `--tts-data-dir`     | Directory for piper voice models (default: `~/.local/share/piper-voices`) |
+| `--tts-cuda`         | Use CUDA GPU acceleration for Piper TTS                                   |
 **Playback timing:**
-| Flag | Description |
-|------|-------------|
-| `--pre-delay` | Seconds to wait after answer before playback (default: 0) |
-| `--post-delay` | Seconds to wait after playback before hangup (default: 0) |
-| `--inter-delay` | Seconds to wait between WAV repeats (default: 0) |
-| `--repeat` | Number of times to play the WAV (default: 1) |
+| Flag                 | Description                                                                                                                                                                     |
+|----------------------|---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
+| `--pre-delay`        | Seconds to wait after answer before playback (default: 0)                                                                                                                       |
+| `--post-delay`       | Seconds to wait after playback before hangup (default: 0)                                                                                                                       |
+| `--inter-delay`      | Seconds to wait between WAV repeats (default: 0)                                                                                                                                |
+| `--repeat`           | Number of times to play the WAV (default: 1)                                                                                                                                    |
 | `--wait-for-silence` | Wait for N seconds of remote silence before playback (e.g. `1.0` to let callee finish "Hello?"). Uses `SilenceDetector` on the incoming audio RMS. Applied after `--pre-delay`. |
 **Recording & transcription:**
-| Flag | Description |
-|------|-------------|
-| `--record` | Record remote-party (RX) audio to this WAV file path (parent dirs created automatically) |
-| `--record-tx` | Record local (TX) audio to this WAV file path |
-| `--mix-mode` | Post-call mix mode: `none`, `mono`, or `stereo` (requires `--record` + `--record-tx`) |
-| `--mix-output` | Output path for the RX+TX mix file |
-| `--transcribe` | Transcribe recorded audio via STT and write a JSON call report (requires `--record`) |
-| `--live-transcribe` | Live STT of remote audio during the call (real-time output to console) |
-| `--stt-backend` | STT backend: `faster-whisper` (default) or `openvino` |
-| `--stt-model` | Whisper model size for transcription (default: `medium`) |
-| `--stt-language` | Language code for STT transcription (default: from config, then `de`) |
-| `--stt-data-dir` | Directory for Whisper STT models (default: `~/.local/share/faster-whisper-models`) |
+| Flag                | Description                                                                              |
+|---------------------|------------------------------------------------------------------------------------------|
+| `--record`          | Record remote-party (RX) audio to this WAV file path (parent dirs created automatically) |
+| `--record-tx`       | Record local (TX) audio to this WAV file path                                            |
+| `--mix-mode`        | Post-call mix mode: `none`, `mono`, or `stereo` (requires `--record` + `--record-tx`)    |
+| `--mix-output`      | Output path for the RX+TX mix file                                                       |
+| `--transcribe`      | Transcribe recorded audio via STT and write a JSON call report (requires `--record`)     |
+| `--live-transcribe` | Live STT of remote audio during the call (real-time output to console)                   |
+| `--stt-backend`     | STT backend: `faster-whisper` (default) or `openvino`                                    |
+| `--stt-model`       | Whisper model size for transcription (default: `medium`)                                 |
+| `--stt-language`    | Language code for STT transcription (default: from config, then `de`)                    |
+| `--stt-data-dir`    | Directory for Whisper STT models (default: `~/.local/share/faster-whisper-models`)       |
 **Live VAD (for `--live-transcribe`):**
-| Flag | Description |
-|------|-------------|
-| `--vad-silence-threshold` | RMS silence threshold (default: 0.01) |
-| `--vad-silence-trigger` | Seconds of silence to trigger chunk boundary (default: 0.3) |
-| `--vad-max-chunk` | Max seconds per audio chunk (default: 5.0) |
-| `--vad-min-chunk` | Min seconds per audio chunk (default: 0.5) |
+| Flag                      | Description                                                 |
+|---------------------------|-------------------------------------------------------------|
+| `--vad-silence-threshold` | RMS silence threshold (default: 0.01)                       |
+| `--vad-silence-trigger`   | Seconds of silence to trigger chunk boundary (default: 0.3) |
+| `--vad-max-chunk`         | Max seconds per audio chunk (default: 5.0)                  |
+| `--vad-min-chunk`         | Min seconds per audio chunk (default: 0.5)                  |
 **Audio device:**
-| Flag | Description |
-|------|-------------|
-| `--no-null-audio` | Use real audio devices instead of null device for both directions |
-| `--real-capture` | Use real microphone even when playback stays null |
-| `--real-playback` | Use real speaker even when capture stays null |
-| `--play-audio` | Play remote-party (RX) audio on local speakers via sounddevice |
-| `--play-tx` | Route local (TX) audio to output sinks; stereo with RX when combined with `--play-audio` |
-| `--no-play-rx` | Disable routing of RX audio to output sinks |
-| `--audio-device` | Sounddevice output device (index or name substring) |
-| `--audio-socket` | Unix socket path for live PCM streaming (16 kHz, S16_LE, mono) |
+| Flag              | Description                                                                              |
+|-------------------|------------------------------------------------------------------------------------------|
+| `--no-null-audio` | Use real audio devices instead of null device for both directions                        |
+| `--real-capture`  | Use real microphone even when playback stays null                                        |
+| `--real-playback` | Use real speaker even when capture stays null                                            |
+| `--play-audio`    | Play remote-party (RX) audio on local speakers via sounddevice                           |
+| `--play-tx`       | Route local (TX) audio to output sinks; stereo with RX when combined with `--play-audio` |
+| `--no-play-rx`    | Disable routing of RX audio to output sinks                                              |
+| `--audio-device`  | Sounddevice output device (index or name substring)                                      |
+| `--audio-socket`  | Unix socket path for live PCM streaming (16 kHz, S16_LE, mono)                           |
 **NAT traversal:**
-| Flag | Description |
-|------|-------------|
-| `--stun-servers` | Comma-separated STUN servers (e.g. `stun.l.google.com:19302`) |
-| `--ice` | Enable ICE for media NAT traversal |
-| `--turn-server` | TURN relay server (`host:port`) |
-| `--turn-username` | TURN username |
-| `--turn-password` | TURN password |
-| `--turn-transport` | TURN transport: `udp`, `tcp`, or `tls` (default: udp) |
-| `--keepalive` | UDP keepalive interval in seconds (0 = disabled) |
-| `--public-address` | Public IP to advertise in SDP/Contact (e.g. K3s node IP) |
+| Flag               | Description                                                   |
+|--------------------|---------------------------------------------------------------|
+| `--stun-servers`   | Comma-separated STUN servers (e.g. `stun.l.google.com:19302`) |
+| `--ice`            | Enable ICE for media NAT traversal                            |
+| `--turn-server`    | TURN relay server (`host:port`)                               |
+| `--turn-username`  | TURN username                                                 |
+| `--turn-password`  | TURN password                                                 |
+| `--turn-transport` | TURN transport: `udp`, `tcp`, or `tls` (default: udp)         |
+| `--keepalive`      | UDP keepalive interval in seconds (0 = disabled)              |
+| `--public-address` | Public IP to advertise in SDP/Contact (e.g. K3s node IP)      |
 **Logging:**
-| Flag | Description |
-|------|-------------|
-| `--verbose`, `-v` | Debug logging |
+| Flag                | Description                           |
+|---------------------|---------------------------------------|
+| `--verbose`, `-v`   | Debug logging                         |
 | `--pjsip-log-level` | PJSIP log verbosity (0–6, default: 3) |
 ### `callee_autoanswer` — Auto-Answer Incoming Calls
@@ -421,23 +425,23 @@ python -m sipstuff.cli callee_autoanswer \
     --answer-delay 2.0
 ```
-| Flag | Description |
-|------|-------------|
-| `--mode` | Playback mode: `none`, `wav`, or `tts` (default: `none`) |
-| `--wav-file` | WAV file to play (requires `--mode wav`) |
-| `--tts-text` | Text for TTS (requires `--mode tts`) |
-| `--piper-model` | Piper voice model (default: `de_DE-thorsten-high`) |
-| `--tts-data-dir` | Piper data directory |
-| `--start-wav` | WAV file to play at call start |
-| `--end-wav` | WAV file to play before hangup |
-| `--pause-before-start` | Pause before start WAV (default: 0.0 s) |
-| `--pause-before-content` | Pause before content WAV/TTS (default: 0.0 s) |
-| `--pause-before-end` | Pause before end WAV (default: 0.0 s) |
-| `--answer-delay` | Seconds before answering (default: 1.0) |
-| `--tts-cuda` | Use CUDA GPU acceleration for Piper TTS |
-| `--no-auto-answer` | Do not auto-answer calls |
-Shared flags: `--config`, `--server`, `--port`, `--user`, `--password`, `--transport`, `--srtp`, `--tls-verify`, `--local-port`, NAT traversal group, `--verbose`, `--pjsip-log-level`
+| Flag                     | Description                                              |
+|--------------------------|----------------------------------------------------------|
+| `--mode`                 | Playback mode: `none`, `wav`, or `tts` (default: `none`) |
+| `--wav-file`             | WAV file to play (requires `--mode wav`)                 |
+| `--tts-text`             | Text for TTS (requires `--mode tts`)                     |
+| `--piper-model`          | Piper voice model (default: `de_DE-thorsten-high`)       |
+| `--tts-data-dir`         | Piper data directory                                     |
+| `--start-wav`            | WAV file to play at call start                           |
+| `--end-wav`              | WAV file to play before hangup                           |
+| `--pause-before-start`   | Pause before start WAV (default: 0.0 s)                  |
+| `--pause-before-content` | Pause before content WAV/TTS (default: 0.0 s)            |
+| `--pause-before-end`     | Pause before end WAV (default: 0.0 s)                    |
+| `--answer-delay`         | Seconds before answering (default: 1.0)                  |
+| `--tts-cuda`             | Use CUDA GPU acceleration for Piper TTS                  |
+| `--no-auto-answer`       | Do not auto-answer calls                                 |
+Shared flags: `--config`, `--server`, `--port`, `--user`, `--password`, `--transport`, `--srtp`, `--tls-verify`, `--rtp-port`, `--rtp-port-range`, `--no-rtp-randomize`, `--local-port`, NAT traversal group, `--verbose`, `--pjsip-log-level`
 ### `callee_realtime-tts` — Incoming Call with Live TTS
@@ -448,67 +452,67 @@ Answer incoming calls and speak text in real-time via Piper TTS:
 python -m sipstuff.cli callee_realtime-tts \
     --server pbx.local --user 1001 --password secret \
     --tts-text "Willkommen, wie kann ich Ihnen helfen?" \
-    --piper-model /path/to/de_DE-thorsten-high.onnx
+    --piper-live-model de_DE-thorsten-high
 # Interactive mode — type text during the call
 python -m sipstuff.cli callee_realtime-tts \
     --server pbx.local --user 1001 --password secret \
     --interactive \
-    --piper-model /path/to/de_DE-thorsten-high.onnx
+    --piper-live-model de_DE-thorsten-high
 ```
 **TTS & playback:**
-| Flag | Description |
-|------|-------------|
-| `--tts-text` | Initial TTS text spoken on answer |
-| `--interactive` | Interactive mode: type text in the console that gets spoken live |
-| `--piper-model` | Path to Piper `.onnx` model (default: `./de_DE-thorsten-high.onnx`) |
-| `--tts-cuda` | Use CUDA GPU acceleration for Piper TTS |
-| `--wav-file` | WAV file to play at call start |
-| `--play-delay` | Seconds to wait before playback (default: 0.0) |
-| `--answer-delay` | Seconds before answering (default: 1.0) |
-| `--no-auto-answer` | Do not auto-answer calls |
+| Flag                 | Description                                                      |
+|----------------------|------------------------------------------------------------------|
+| `--tts-text`         | Initial TTS text spoken on answer                                |
+| `--interactive`      | Interactive mode: type text in the console that gets spoken live |
+| `--piper-live-model` | Piper voice model (default: `de_DE-thorsten-high`)               |
+| `--tts-cuda`         | Use CUDA GPU acceleration for Piper TTS                          |
+| `--wav-file`         | WAV file to play at call start                                   |
+| `--play-delay`       | Seconds to wait before playback (default: 0.0)                   |
+| `--answer-delay`     | Seconds before answering (default: 1.0)                          |
+| `--no-auto-answer`   | Do not auto-answer calls                                         |
 **STT (speech-to-text):**
-| Flag | Description |
-|------|-------------|
-| `--stt-backend` | STT backend: `faster-whisper` (default) or `openvino` |
-| `--stt-model` | Whisper model size or HuggingFace model ID (default: `base`) |
-| `--stt-live-model` | Separate smaller model for live transcription |
-| `--stt-device` | Compute device: `cpu` or `cuda` |
-| `--stt-language` | Language code for STT |
-| `--stt-data-dir` | Whisper model cache directory |
+| Flag               | Description                                                  |
+|--------------------|--------------------------------------------------------------|
+| `--stt-backend`    | STT backend: `faster-whisper` (default) or `openvino`        |
+| `--stt-model`      | Whisper model size or HuggingFace model ID (default: `base`) |
+| `--stt-live-model` | Separate smaller model for live transcription                |
+| `--stt-device`     | Compute device: `cpu` or `cuda`                              |
+| `--stt-language`   | Language code for STT                                        |
+| `--stt-data-dir`   | Whisper model cache directory                                |
 **Live VAD:**
-| Flag | Description |
-|------|-------------|
-| `--vad-silence-threshold` | RMS silence threshold (default: 0.01) |
-| `--vad-silence-trigger` | Seconds of silence to trigger chunk boundary (default: 0.3) |
-| `--vad-max-chunk` | Max seconds per audio chunk (default: 5.0) |
-| `--vad-min-chunk` | Min seconds per audio chunk (default: 0.5) |
+| Flag                      | Description                                                 |
+|---------------------------|-------------------------------------------------------------|
+| `--vad-silence-threshold` | RMS silence threshold (default: 0.01)                       |
+| `--vad-silence-trigger`   | Seconds of silence to trigger chunk boundary (default: 0.3) |
+| `--vad-max-chunk`         | Max seconds per audio chunk (default: 5.0)                  |
+| `--vad-min-chunk`         | Min seconds per audio chunk (default: 0.5)                  |
 **Recording & transcription:**
-| Flag | Description |
-|------|-------------|
-| `--wav-output` | Save RX recording to this path |
-| `--wav-dir` | Directory for WAV files (default: `..`) |
-| `--wav-output-tx` | Save TX recording to this path |
-| `--no-wav` | Do not save WAV recordings |
-| `--transcribe` | Full transcription of RX recording after call ends (writes JSON report) |
+| Flag              | Description                                                             |
+|-------------------|-------------------------------------------------------------------------|
+| `--wav-output`    | Save RX recording to this path                                          |
+| `--wav-dir`       | Directory for WAV files (default: `..`)                                 |
+| `--wav-output-tx` | Save TX recording to this path                                          |
+| `--no-wav`        | Do not save WAV recordings                                              |
+| `--transcribe`    | Full transcription of RX recording after call ends (writes JSON report) |
 **Audio output:**
-| Flag | Description |
-|------|-------------|
+| Flag             | Description                                                     |
+|------------------|-----------------------------------------------------------------|
 | `--audio-socket` | Unix socket for live audio streaming (PCM 16 kHz, S16_LE, mono) |
-| `--play-audio` | Play remote audio on local speakers via sounddevice |
-| `--audio-device` | Sounddevice output device (index or name substring) |
+| `--play-audio`   | Play remote audio on local speakers via sounddevice             |
+| `--audio-device` | Sounddevice output device (index or name substring)             |
-Shared flags: `--config`, `--server`, `--port`, `--user`, `--password`, `--transport`, `--srtp`, `--tls-verify`, `--local-port`, NAT traversal group, `--verbose`, `--pjsip-log-level`
+Shared flags: `--config`, `--server`, `--port`, `--user`, `--password`, `--transport`, `--srtp`, `--tls-verify`, `--rtp-port`, `--rtp-port-range`, `--no-rtp-randomize`, `--local-port`, NAT traversal group, `--verbose`, `--pjsip-log-level`
 ### `callee_live-transcribe` — Incoming Call with Live Transcription
@@ -533,35 +537,35 @@ python -m sipstuff.cli callee_live-transcribe \
     --audio-socket /tmp/audio.sock --play-audio
 ```
-| Flag | Description |
-|------|-------------|
-| `--stt-backend` | STT backend: `faster-whisper` (default) or `openvino` |
-| `--stt-model` | Whisper model (default: `base`) |
-| `--stt-live-model` | Separate smaller model for live transcription |
-| `--stt-device` | Compute device: `cpu` or `cuda` |
-| `--stt-language` | Language code for STT |
-| `--stt-data-dir` | Whisper model cache directory |
-| `--vad-silence-threshold` | RMS silence threshold (default: 0.01) |
-| `--vad-silence-trigger` | Seconds of silence to trigger chunk boundary (default: 0.3) |
-| `--vad-max-chunk` | Max seconds per audio chunk (default: 5.0) |
-| `--vad-min-chunk` | Min seconds per audio chunk (default: 0.5) |
-| `--wav-output` | Save RX recording to this path |
-| `--wav-output-tx` | Save TX recording to this path |
-| `--wav-dir` | Directory for WAV files (default: `..`) |
-| `--no-wav` | Do not save WAV recordings |
-| `--transcribe` | Full transcription of RX recording after call ends (writes JSON report) |
-| `--wav-file` | WAV file to play at call start |
-| `--tts-text` | Text for Piper TTS playback at call start |
-| `--piper-model` | Piper voice model (default: `de_DE-thorsten-high`) |
-| `--tts-data-dir` | Piper data directory |
-| `--play-delay` | Seconds to wait before playback (default: 0.0) |
-| `--audio-socket` | Unix socket for live audio streaming (PCM 16 kHz, S16_LE, mono) |
-| `--play-audio` | Play remote audio on local speakers via sounddevice |
-| `--audio-device` | Sounddevice output device (index or name substring) |
-| `--answer-delay` | Seconds before answering (default: 1.0) |
-| `--no-auto-answer` | Do not auto-answer calls |
-Shared flags: `--config`, `--server`, `--port`, `--user`, `--password`, `--transport`, `--srtp`, `--tls-verify`, `--local-port`, NAT traversal group, `--verbose`, `--pjsip-log-level`
+| Flag                      | Description                                                             |
+|---------------------------|-------------------------------------------------------------------------|
+| `--stt-backend`           | STT backend: `faster-whisper` (default) or `openvino`                   |
+| `--stt-model`             | Whisper model (default: `base`)                                         |
+| `--stt-live-model`        | Separate smaller model for live transcription                           |
+| `--stt-device`            | Compute device: `cpu` or `cuda`                                         |
+| `--stt-language`          | Language code for STT                                                   |
+| `--stt-data-dir`          | Whisper model cache directory                                           |
+| `--vad-silence-threshold` | RMS silence threshold (default: 0.01)                                   |
+| `--vad-silence-trigger`   | Seconds of silence to trigger chunk boundary (default: 0.3)             |
+| `--vad-max-chunk`         | Max seconds per audio chunk (default: 5.0)                              |
+| `--vad-min-chunk`         | Min seconds per audio chunk (default: 0.5)                              |
+| `--wav-output`            | Save RX recording to this path                                          |
+| `--wav-output-tx`         | Save TX recording to this path                                          |
+| `--wav-dir`               | Directory for WAV files (default: `..`)                                 |
+| `--no-wav`                | Do not save WAV recordings                                              |
+| `--transcribe`            | Full transcription of RX recording after call ends (writes JSON report) |
+| `--wav-file`              | WAV file to play at call start                                          |
+| `--tts-text`              | Text for Piper TTS playback at call start                               |
+| `--piper-model`           | Piper voice model (default: `de_DE-thorsten-high`)                      |
+| `--tts-data-dir`          | Piper data directory                                                    |
+| `--play-delay`            | Seconds to wait before playback (default: 0.0)                          |
+| `--audio-socket`          | Unix socket for live audio streaming (PCM 16 kHz, S16_LE, mono)         |
+| `--play-audio`            | Play remote audio on local speakers via sounddevice                     |
+| `--audio-device`          | Sounddevice output device (index or name substring)                     |
+| `--answer-delay`          | Seconds before answering (default: 1.0)                                 |
+| `--no-auto-answer`        | Do not auto-answer calls                                                |
+Shared flags: `--config`, `--server`, `--port`, `--user`, `--password`, `--transport`, `--srtp`, `--tls-verify`, `--rtp-port`, `--rtp-port-range`, `--no-rtp-randomize`, `--local-port`, NAT traversal group, `--verbose`, `--pjsip-log-level`
 ## Docker / Podman Example
@@ -667,11 +671,11 @@ Notes:
 <video src="https://github.com/user-attachments/assets/996c63d2-4f40-4402-9107-9d8b4b8827eb" width="100%" controls></video>
-![Call startup — TTS generation, SIP registration, silence detection, and WAV playback](Bildschirmfoto_2026-02-15_17-40-19_blurred.png)
+![Call startup — TTS generation, SIP registration, silence detection, and WAV playback](https://raw.githubusercontent.com/vroomfondel/sipstuff/main/Bildschirmfoto_2026-02-15_17-40-19_blurred.png)
-![Call completion — RTP stats, STT transcription, and JSON call report](Bildschirmfoto_2026-02-15_17-40-43_blurred.png)
+![Call completion — RTP stats, STT transcription, and JSON call report](https://raw.githubusercontent.com/vroomfondel/sipstuff/main/Bildschirmfoto_2026-02-15_17-40-43_blurred.png)
-![JSON call report detail and follow-up call](Bildschirmfoto_2026-02-15_17-41-02_blurred.png)
+![JSON call report detail and follow-up call](https://raw.githubusercontent.com/vroomfondel/sipstuff/main/Bildschirmfoto_2026-02-15_17-41-02_blurred.png)
 ## Library Usage
@@ -850,6 +854,9 @@ sip:
   srtp: "disabled"        # disabled, optional, or mandatory
   tls_verify_server: false
   local_port: 0           # 0 = auto
+  rtp_port: 4000          # 0 = OS-assigned
+  rtp_port_range: 200
+  rtp_randomize_port: true
 call:
   timeout: 60
@@ -915,111 +922,114 @@ All settings can be set via `SIP_` prefixed environment variables:
 **SIP connection:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_SERVER` | `sip.server` |
-| `SIP_PORT` | `sip.port` |
-| `SIP_USER` | `sip.user` |
-| `SIP_PASSWORD` | `sip.password` |
-| `SIP_TRANSPORT` | `sip.transport` |
-| `SIP_SRTP` | `sip.srtp` |
-| `SIP_TLS_VERIFY_SERVER` | `sip.tls_verify_server` |
-| `SIP_LOCAL_PORT` | `sip.local_port` |
+| Variable                 | Maps to                  |
+|--------------------------|--------------------------|
+| `SIP_SERVER`             | `sip.server`             |
+| `SIP_PORT`               | `sip.port`               |
+| `SIP_USER`               | `sip.user`               |
+| `SIP_PASSWORD`           | `sip.password`           |
+| `SIP_TRANSPORT`          | `sip.transport`          |
+| `SIP_SRTP`               | `sip.srtp`               |
+| `SIP_TLS_VERIFY_SERVER`  | `sip.tls_verify_server`  |
+| `SIP_LOCAL_PORT`         | `sip.local_port`         |
+| `SIP_RTP_PORT`           | `sip.rtp_port`           |
+| `SIP_RTP_PORT_RANGE`     | `sip.rtp_port_range`     |
+| `SIP_RTP_RANDOMIZE_PORT` | `sip.rtp_randomize_port` |
 **Call timing:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_TIMEOUT` | `call.timeout` |
-| `SIP_PRE_DELAY` | `call.pre_delay` |
-| `SIP_POST_DELAY` | `call.post_delay` |
-| `SIP_INTER_DELAY` | `call.inter_delay` |
-| `SIP_REPEAT` | `call.repeat` |
+| Variable               | Maps to                 |
+|------------------------|-------------------------|
+| `SIP_TIMEOUT`          | `call.timeout`          |
+| `SIP_PRE_DELAY`        | `call.pre_delay`        |
+| `SIP_POST_DELAY`       | `call.post_delay`       |
+| `SIP_INTER_DELAY`      | `call.inter_delay`      |
+| `SIP_REPEAT`           | `call.repeat`           |
 | `SIP_WAIT_FOR_SILENCE` | `call.wait_for_silence` |
 **TTS:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_TTS_MODEL` | `tts.model` |
+| Variable              | Maps to           |
+|-----------------------|-------------------|
+| `SIP_TTS_MODEL`       | `tts.model`       |
 | `SIP_TTS_SAMPLE_RATE` | `tts.sample_rate` |
-| `SIP_TTS_CUDA` | `tts.use_cuda` |
-| `SIP_TTS_DATA_DIR` | `tts.data_dir` |
+| `SIP_TTS_CUDA`        | `tts.use_cuda`    |
+| `SIP_TTS_DATA_DIR`    | `tts.data_dir`    |
 **NAT traversal:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_STUN_SERVERS` | `nat.stun_servers` (comma-separated) |
-| `SIP_STUN_IGNORE_FAILURE` | `nat.stun_ignore_failure` |
-| `SIP_ICE_ENABLED` | `nat.ice_enabled` |
-| `SIP_TURN_ENABLED` | `nat.turn_enabled` |
-| `SIP_TURN_SERVER` | `nat.turn_server` |
-| `SIP_TURN_USERNAME` | `nat.turn_username` |
-| `SIP_TURN_PASSWORD` | `nat.turn_password` |
-| `SIP_TURN_TRANSPORT` | `nat.turn_transport` |
-| `SIP_KEEPALIVE_SEC` | `nat.keepalive_sec` |
-| `SIP_PUBLIC_ADDRESS` | `nat.public_address` |
+| Variable                  | Maps to                              |
+|---------------------------|--------------------------------------|
+| `SIP_STUN_SERVERS`        | `nat.stun_servers` (comma-separated) |
+| `SIP_STUN_IGNORE_FAILURE` | `nat.stun_ignore_failure`            |
+| `SIP_ICE_ENABLED`         | `nat.ice_enabled`                    |
+| `SIP_TURN_ENABLED`        | `nat.turn_enabled`                   |
+| `SIP_TURN_SERVER`         | `nat.turn_server`                    |
+| `SIP_TURN_USERNAME`       | `nat.turn_username`                  |
+| `SIP_TURN_PASSWORD`       | `nat.turn_password`                  |
+| `SIP_TURN_TRANSPORT`      | `nat.turn_transport`                 |
+| `SIP_KEEPALIVE_SEC`       | `nat.keepalive_sec`                  |
+| `SIP_PUBLIC_ADDRESS`      | `nat.public_address`                 |
 **Audio device:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_NULL_AUDIO` | `audio.use_null_audio` (default: `true`) |
-| `SIP_NULL_CAPTURE` | `audio.null_capture` (null mic; `None` inherits `SIP_NULL_AUDIO`) |
+| Variable            | Maps to                                                                |
+|---------------------|------------------------------------------------------------------------|
+| `SIP_NULL_AUDIO`    | `audio.use_null_audio` (default: `true`)                               |
+| `SIP_NULL_CAPTURE`  | `audio.null_capture` (null mic; `None` inherits `SIP_NULL_AUDIO`)      |
 | `SIP_NULL_PLAYBACK` | `audio.null_playback` (null speaker; `None` inherits `SIP_NULL_AUDIO`) |
-| `SIP_PLAY_RX` | `audio.play_rx` (route RX to output sinks; default: `true`) |
-| `SIP_PLAY_TX` | `audio.play_tx` (route TX to output sinks; default: `false`) |
-| `SIP_AUDIO_DEVICE` | `audio.audio_device` (sounddevice index or name) |
+| `SIP_PLAY_RX`       | `audio.play_rx` (route RX to output sinks; default: `true`)            |
+| `SIP_PLAY_TX`       | `audio.play_tx` (route TX to output sinks; default: `false`)           |
+| `SIP_AUDIO_DEVICE`  | `audio.audio_device` (sounddevice index or name)                       |
 **STT:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_STT_BACKEND` | `stt.backend` |
-| `SIP_STT_MODEL` | `stt.model` |
-| `SIP_STT_LANGUAGE` | `stt.language` |
-| `SIP_STT_DEVICE` | `stt.device` |
-| `SIP_STT_DATA_DIR` | `stt.data_dir` |
+| Variable              | Maps to               |
+|-----------------------|-----------------------|
+| `SIP_STT_BACKEND`     | `stt.backend`         |
+| `SIP_STT_MODEL`       | `stt.model`           |
+| `SIP_STT_LANGUAGE`    | `stt.language`        |
+| `SIP_STT_DEVICE`      | `stt.device`          |
+| `SIP_STT_DATA_DIR`    | `stt.data_dir`        |
 | `SIP_LIVE_TRANSCRIBE` | `stt.live_transcribe` |
 **VAD:**
-| Variable | Maps to |
-|----------|---------|
+| Variable                    | Maps to                 |
+|-----------------------------|-------------------------|
 | `SIP_VAD_SILENCE_THRESHOLD` | `vad.silence_threshold` |
-| `SIP_VAD_SILENCE_TRIGGER` | `vad.silence_trigger` |
-| `SIP_VAD_MAX_CHUNK` | `vad.max_chunk` |
-| `SIP_VAD_MIN_CHUNK` | `vad.min_chunk` |
+| `SIP_VAD_SILENCE_TRIGGER`   | `vad.silence_trigger`   |
+| `SIP_VAD_MAX_CHUNK`         | `vad.max_chunk`         |
+| `SIP_VAD_MIN_CHUNK`         | `vad.min_chunk`         |
 **Callee:**
-| Variable | Maps to |
-|----------|---------|
-| `SIP_AUTO_ANSWER` | `callee.auto_answer` (default: `true`) |
+| Variable           | Maps to                                |
+|--------------------|----------------------------------------|
+| `SIP_AUTO_ANSWER`  | `callee.auto_answer` (default: `true`) |
 | `SIP_ANSWER_DELAY` | `callee.answer_delay` (default: `1.0`) |
 **TTS runtime:**
-| Variable | Default | Description |
-|----------|---------|-------------|
+| Variable         | Default                       | Description                           |
+|------------------|-------------------------------|---------------------------------------|
 | `PIPER_DATA_DIR` | `~/.local/share/piper-voices` | Directory for downloaded voice models |
 **STT runtime (faster-whisper):**
-| Variable | Default | Description |
-|----------|---------|-------------|
-| `WHISPER_DATA_DIR` | `~/.local/share/faster-whisper-models` | Directory for downloaded Whisper models |
-| `WHISPER_MODEL` | `medium` | Default model size (`tiny`, `base`, `small`, `medium`, `large-v3`) |
-| `WHISPER_DEVICE` | `cpu` | Compute device (`cpu` or `cuda`) |
-| `WHISPER_COMPUTE_TYPE` | `int8` (CPU) / `float16` (CUDA) | Quantization type (`int8`, `float16`, `float32`) |
+| Variable               | Default                                | Description                                                        |
+|------------------------|----------------------------------------|--------------------------------------------------------------------|
+| `WHISPER_DATA_DIR`     | `~/.local/share/faster-whisper-models` | Directory for downloaded Whisper models                            |
+| `WHISPER_MODEL`        | `medium`                               | Default model size (`tiny`, `base`, `small`, `medium`, `large-v3`) |
+| `WHISPER_DEVICE`       | `cpu`                                  | Compute device (`cpu` or `cuda`)                                   |
+| `WHISPER_COMPUTE_TYPE` | `int8` (CPU) / `float16` (CUDA)        | Quantization type (`int8`, `float16`, `float32`)                   |
 **PJSIP native log routing:**
-| Variable | Default | Description |
-|----------|---------|-------------|
-| `PJSIP_LOG_LEVEL` | `3` | PJSIP log verbosity routed through loguru (0 = none … 6 = trace) |
-| `PJSIP_CONSOLE_LEVEL` | `4` | PJSIP native console output printed directly to stdout (4 = PJSIP default, 0 = suppressed) |
+| Variable              | Default | Description                                                                                |
+|-----------------------|---------|--------------------------------------------------------------------------------------------|
+| `PJSIP_LOG_LEVEL`     | `3`     | PJSIP log verbosity routed through loguru (0 = none … 6 = trace)                           |
+| `PJSIP_CONSOLE_LEVEL` | `4`     | PJSIP native console output printed directly to stdout (4 = PJSIP default, 0 = suppressed) |
 All PJSIP native output is captured by a `pj.LogWriter` subclass and forwarded to loguru (`classname="pjsip"`). `PJSIP_LOG_LEVEL` controls what the writer receives; `PJSIP_CONSOLE_LEVEL` controls what PJSIP additionally prints to stdout on its own. Set `PJSIP_CONSOLE_LEVEL=0` to suppress native output and rely solely on loguru.
@@ -1061,28 +1071,30 @@ Non-standard formats (stereo, different bit depths/rates) will produce warnings
 ## Module Structure
-| File | Purpose |
-|------|---------|
-| `__init__.py` | Public API: `make_sip_call`, `SipCaller`, `SipCallee`, `SipEndpoint`, `SipCallError`, `CallResult`, config classes, `generate_wav`, `transcribe_wav`, `configure_logging`, `print_banner` |
-| `sip_endpoint.py` | `SipEndpoint` base class for PJSUA2 Endpoint lifecycle, `SipCaller(SipEndpoint)` with `make_call()` orchestration, `_PjLogWriter`, `_local_address_for()` |
-| `sip_call.py` | `SipCall(pj.Call)` shared base, `SipCallerCall` (caller-side media management), `SipCalleeCall` (callee-side hooks) |
-| `sip_media.py` | `SilenceDetector`, `AudioStreamPort`, `TranscriptionPort` — PJSUA2 `AudioMediaPort` subclasses |
-| `sip_account.py` | `SipAccount(pj.Account)` base, `SipCallerAccount` (incoming-call rejection), `SipCalleeAccount` (auto-answer dispatch) |
-| `sip_callee.py` | `SipCallee(SipEndpoint)` — incoming call handling orchestration |
-| `sip_types.py` | `CallResult` dataclass, `SipCallError`, `WavInfo` |
-| `sip_caller.py` | Legacy module (retained for imports) |
-| `sipconfig.py` | Pydantic v2 config models (`SipEndpointConfig`, `SipCallerConfig`, `SipCalleeConfig`) with YAML / env / override loading |
-| `vad.py` | `VADAudioBuffer` — voice activity detection buffer for live transcription |
-| `audio.py` | `resample_linear()` (numpy), `ensure_wav_16k_mono()` — audio format utilities |
-| `snddevice_list.py` | Sound device enumeration utility |
-| `tts/tts.py` | Piper TTS integration: text-to-WAV generation via Python API with optional resampling |
-| `tts/live.py` | Live TTS streaming: `PiperTTSProducer` (producer thread), `TTSMediaPort` (PJSIP consumer), `interactive_console()` |
-| `stt/stt.py` | Speech-to-text via faster-whisper: WAV transcription with Silero VAD pre-filtering and segment timestamps |
-| `stt/live.py` | Live STT streaming during calls |
-| `autoanswer/` | Callee auto-answer implementation (integrated as `callee_autoanswer` CLI subcommand) |
-| `realtime/` | Callee real-time TTS implementation (integrated as `callee_realtime-tts` CLI subcommand) |
-| `transcribe/` | Callee live-transcribe implementation (integrated as `callee_live-transcribe` CLI subcommand) |
-| `training/` | Voice training utilities: `record_dataset.py`, `record_gui.py` (standalone, not in CLI) |
+| File                | Purpose                                                                                                                                                                                   |
+|---------------------|-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
+| `__init__.py`       | Public API: `make_sip_call`, `SipCaller`, `SipCallee`, `SipEndpoint`, `SipCallError`, `CallResult`, config classes, `generate_wav`, `transcribe_wav`, `configure_logging`, `print_banner` |
+| `sip_endpoint.py`   | `SipEndpoint` base class for PJSUA2 Endpoint lifecycle, `SipCaller(SipEndpoint)` with `make_call()` orchestration, `_PjLogWriter`, `_local_address_for()`                                 |
+| `sip_call.py`       | `SipCall(pj.Call)` shared base, `SipCallerCall` (caller-side media management), `SipCalleeCall` (callee-side hooks)                                                                       |
+| `sip_media.py`      | `SilenceDetector`, `AudioStreamPort`, `TranscriptionPort` — PJSUA2 `AudioMediaPort` subclasses                                                                                            |
+| `sip_account.py`    | `SipAccount(pj.Account)` base, `SipCallerAccount` (incoming-call rejection), `SipCalleeAccount` (auto-answer dispatch)                                                                    |
+| `sip_callee.py`     | `SipCallee(SipEndpoint)` — incoming call handling orchestration                                                                                                                           |
+| `sip_types.py`      | `CallResult` dataclass, `SipCallError`, `WavInfo`                                                                                                                                         |
+| `sip_caller.py`     | Legacy module (retained for imports)                                                                                                                                                      |
+| `sipconfig.py`      | Pydantic v2 config models (`SipEndpointConfig`, `SipCallerConfig`, `SipCalleeConfig`) with YAML / env / override loading                                                                  |
+| `vad.py`            | `VADAudioBuffer` — voice activity detection buffer for live transcription                                                                                                                 |
+| `audio.py`          | `resample_linear()` (numpy), `ensure_wav_16k_mono()` — audio format utilities                                                                                                             |
+| `snddevice_list.py` | Sound device enumeration utility                                                                                                                                                          |
+| `tts/tts.py`        | Piper TTS integration: text-to-WAV generation via Python API with optional resampling                                                                                                     |
+| `tts/live.py`       | Live TTS streaming: `PiperTTSProducer` (producer thread), `TTSMediaPort` (PJSIP consumer), `interactive_console()`                                                                        |
+| `stt/stt.py`        | Speech-to-text via faster-whisper: WAV transcription with Silero VAD pre-filtering and segment timestamps                                                                                 |
+| `stt/live.py`       | Live STT streaming during calls                                                                                                                                                           |
+| `autoanswer/`       | Callee auto-answer implementation (integrated as `callee_autoanswer` CLI subcommand)                                                                                                      |
+| `realtime/`         | Callee real-time TTS implementation (integrated as `callee_realtime-tts` CLI subcommand)                                                                                                  |
+| `transcribe/`       | Callee live-transcribe implementation (integrated as `callee_live-transcribe` CLI subcommand)                                                                                             |
+| `training/`         | Voice training utilities: `record_dataset.py`, `record_gui.py` (standalone, not in CLI) — see [training README](https://github.com/vroomfondel/sipstuff/blob/main/sipstuff/training/README.md)                                              |
+![Audio device enumeration (sounddevice input/output devices)](https://raw.githubusercontent.com/vroomfondel/sipstuff/main/Bildschirmfoto_2026-02-22_12-08-06.png)
 | `cli.py` | CLI entry point with six subcommands |
 | `example_config.yaml` | Sample configuration file |
 | `dist_scripts/install_pjsip.sh` | Build script for PJSIP with Python bindings (default: 2.16) |
@@ -1136,5 +1148,18 @@ Tests cover config validation (SipConfig, CallConfig, SipCallerConfig, NatConfig
 CI skips venv setup when `GITHUB_RUN_ID` is set — installs deps directly.
 ## License
+This project is licensed under the LGPL where applicable/possible — see [LICENSE.md](https://github.com/vroomfondel/sipstuff/blob/main/LICENSE.md). Some files/parts may use other licenses: [MIT](https://github.com/vroomfondel/sipstuff/blob/main/LICENSEMIT.md) | [GPL](https://github.com/vroomfondel/sipstuff/blob/main/LICENSEGPL.md) | [LGPL](https://github.com/vroomfondel/sipstuff/blob/main/LICENSELGPL.md). Always check per‑file headers/comments.
+## Authors
+- Repo owner (primary author)
+- Additional attributions are noted inline in code comments
+## Acknowledgments
+- Inspirations and snippets are referenced in code comments where appropriate.
+## ⚠️ Note
-LGPLv3 — see [LICENSE.md](LICENSE.md), [LICENSELGPL.md](LICENSELGPL.md), [LICENSEGPL.md](LICENSEGPL.md), [LICENSEMIT.md](LICENSEMIT.md).
+This is a development/experimental project. For production use, review security settings, customize configurations, and test thoroughly in your environment. Provided "as is" without warranty of any kind, express or implied, including but not limited to the warranties of merchantability, fitness for a particular purpose and noninfringement. In no event shall the authors or copyright holders be liable for any claim, damages or other liability, whether in an action of contract, tort or otherwise, arising from, out of or in connection with the software or the use or other dealings in the software. Use at your own risk.

sipstuff 0.0.2__tar.gz → 0.0.4__tar.gz

sipstuff 0.0.2tar.gz → 0.0.4tar.gz