# mod_audio_stream **Repository Path**: derekmy/mod_audio_stream ## Basic Information - **Project Name**: mod_audio_stream - **Description**: No description available - **Primary Language**: C - **License**: MIT - **Default Branch**: main - **Homepage**: None - **GVP Project**: No ## Statistics - **Stars**: 0 - **Forks**: 1 - **Created**: 2025-11-28 - **Last Updated**: 2025-11-28 ## Categories & Tags **Categories**: Uncategorized **Tags**: None ## README # mod_audio_stream ## Installation ### Dependencies It requires `libfreeswitch-dev`, `libssl-dev`, `zlib1g-dev`, `libevent-dev` and `libspeexdsp-dev` on Debian/Ubuntu which are regular packages for Freeswitch installation. ### Building After cloning please execute: **git submodule init** and **git submodule update** to initialize the submodule. #### Custom path If you built FreeSWITCH from source, eq. install dir is /usr/local/freeswitch, add path to pkgconfig: ``` apt -y install cmake apt -y install libfreeswitch-dev libssl-dev zlib1g-dev libevent-dev libspeexdsp-dev export PKG_CONFIG_PATH=/usr/local/freeswitch/lib/pkgconfig git submodule init git submodule update mkdir build && cd build cmake -DUSE_TLS=ON -DCMAKE_BUILD_TYPE=Release .. make make install ``` #### DEB Package 要构建DEB包,在'make'后构建DEB包，请执行以下操作： ``` make cpack -G DEB ``` Debian软件包将放置在/root/_packages文件夹中。 ## Scripted Build & Installation ``` sudo apt-get -y install git \ && cd /usr/src/ \ && git clone https://github.com/amigniter/mod_audio_stream.git \ && cd mod_audio_stream \ && sudo bash ./build-mod-audio-stream.sh ``` ### Channel variables The following channel variables can be used to fine tune websocket connection and also configure mod_audio_stream logging: | Variable | Description | Default | | -------------------------------------- | ------------------------------------------------------- | ------- | | STREAM_MESSAGE_DEFLATE | true or 1, disables per message deflate | off | | STREAM_HEART_BEAT | number of seconds, interval to send the heart beat | off | | STREAM_SUPPRESS_LOG | true or 1, suppresses printing to log | off | | STREAM_BUFFER_SIZE | buffer duration in milliseconds, divisible by 20 | 20 | | STREAM_EXTRA_HEADERS | JSON object for additional headers in string format | none | | ~~STREAM_NO_RECONNECT~~ | true or 1, disables automatic websocket reconnection | off | | STREAM_TLS_CA_FILE | CA cert or bundle, or the special values SYSTEM or NONE | SYSTEM | | STREAM_TLS_KEY_FILE | optional client key for WSS connections | none | | STREAM_TLS_CERT_FILE | optional client cert for WSS connections | none | | STREAM_TLS_DISABLE_HOSTNAME_VALIDATION | true or 1 disable hostname check in WSS connections | false | - Per message deflate compression option is enabled by default. It can lead to a very nice bandwidth savings. To disable it set the channel var to `true|1`. - Heart beat, sent every xx seconds when there is no traffic to make sure that load balancers do not kill an idle connection. - Suppress parameter is omitted by default(false). All the responses from websocket server will be printed to the log. Not to flood the log you can suppress it by setting the value to `true|1`. Events are fired still, it only affects printing to the log. - `Buffer Size` actually represents a duration of audio chunk sent to websocket. If you want to send e.g. 100ms audio packets to your ws endpoint you would set this variable to 100. If ommited, default packet size of 20ms will be sent as grabbed from the audio channel (which is default FreeSWITCH frame size) - Extra headers should be a JSON object with key-value pairs representing additional HTTP headers. Each key should be a header name, and its corresponding value should be a string. ```json { "Header1": "Value1", "Header2": "Value2", "Header3": "Value3" } - ~~Websocket automatic reconnection is on by default. To disable it set this channel variable to true or 1.~~ - libwsc does not support automatic reconnection. - TLS (for WSS) options can be fine tuned with the `STREAM_TLS_*` channel variables: - `STREAM_TLS_CA_FILE` the ca certificate (or certificate bundle) file. By default is `SYSTEM` which means use the system defaults. Can be `NONE` which result in no peer verification. - `STREAM_TLS_CERT_FILE` optional client tls certificate file sent to the server. - `STREAM_TLS_KEY_FILE` optional client tls key file for the given certificate. - `STREAM_TLS_DISABLE_HOSTNAME_VALIDATION` if `true`, disables the check of the hostname against the peer server certificate. Defaults to `false`, which enforces hostname match with the peer certificate. ## API ### Commands The freeswitch module exposes the following API commands: ``` uuid_audio_stream start ``` Attaches a media bug and starts streaming audio (in L16 format) to the websocket server. FS default is 8k. If sampling-rate is other than 8k it will be resampled. - `uuid` - Freeswitch channel unique id - `wss-url` - websocket url `ws://` or `wss://` - `mix-type` - choice of - "mono" - single channel containing caller's audio - "mixed" - single channel containing both caller and callee audio - "stereo" - two channels with caller audio in one and callee audio in the other. - `sampling-rate` - choice of - "8k" = 8000 Hz sample rate will be generated - "16k" = 16000 Hz sample rate will be generated - `metadata` - (optional) a valid `utf-8` text to send. It will be sent the first before audio streaming starts. ``` uuid_audio_stream send_text ``` Sends a text to the websocket server. Requires a valid `utf-8` text. ``` uuid_audio_stream stop ``` Stops audio stream and closes websocket connection. If _metadata_ is provided it will be sent before the connection is closed. ``` uuid_audio_stream pause ``` Pauses audio stream ``` uuid_audio_stream resume ``` Resumes audio stream ## Events Module will generate the following event types: - `mod_audio_stream::json` - `mod_audio_stream::connect` - `mod_audio_stream::disconnect` - `mod_audio_stream::error` - `mod_audio_stream::play` ### response Message received from websocket endpoint. Json expected, but it contains whatever the websocket server's response is. #### Freeswitch event generated **Name**: mod_audio_stream::json **Body**: WebSocket server response ### connect Successfully connected to websocket server. #### Freeswitch event generated **Name**: mod_audio_stream::connect **Body**: JSON ```json { "status": "connected" } ``` ### disconnect Disconnected from websocket server. #### Freeswitch event generated **Name**: mod_audio_stream::disconnect **Body**: JSON ```json { "status": "disconnected", "message": { "code": 1000, "reason": "Normal closure" } } ``` - code: `` - reason: `` ### error There is an error with the connection. Multiple fields will be available on the event to describe the error. #### Freeswitch event generated **Name**: mod_audio_stream::error **Body**: JSON ```json { "status": "error", "message": { "code": 1, "error": "String explaining the error" } } ``` - code: `` - error: `` #### Possible `code` values | Code | Enum Name | Meaning | |:----:|:----------------------|:-----------------------------------------------------| | 1 | `IO` | I/O error when reading/writing sockets | | 2 | `INVALID_HEADER` | Server sent a malformed WebSocket header | | 3 | `SERVER_MASKED` | Server frames were masked (not allowed by spec) | | 4 | `NOT_SUPPORTED` | Requested feature (e.g. extension) not supported | | 5 | `PING_TIMEOUT` | No PONG received within timeout | | 6 | `CONNECT_FAILED` | TCP connection or DNS lookup failed | | 7 | `TLS_INIT_FAILED` | Couldn't initialize SSL/TLS context | | 8 | `SSL_HANDSHAKE_FAILED`| SSL/TLS handshake with server failed | | 9 | `SSL_ERROR` | Generic OpenSSL error (certificate, cipher, etc.) | ### play **Name**: mod_audio_stream::play **Body**: JSON Websocket server may return JSON object containing base64 encoded audio to be played by the user. To use this feature, response must follow the format: ```json { "type": "streamAudio", "data": { "audioDataType": "raw", "sampleRate": 8000, "audioData": "base64 encoded audio" } } ``` - audioDataType: `` Event generated by the module (subclass: _mod_audio_stream::play_) will be the same as the `data` element with the **file** added to it representing filePath: ```json { "audioDataType": "raw", "sampleRate": 8000, "file": "/path/to/the/file" } ``` If printing to the log is not suppressed, `response` printed to the console will look the same as the event. The original response containing base64 encoded audio is replaced because it can be quite huge. All the files generated by this feature will reside at the temp directory and will be deleted when the session is closed.