### Load configuration from a JSON file

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

This example shows how to start the webrtc-streamer using a configuration file specified by the -C option.

```bash
./webrtc-streamer -C config.json
```

--------------------------------

### Download WebRTC Source and Install Sysroot

Source: https://github.com/mpromonet/webrtc-streamer/wiki/Cross-compile-WebRTC-for-Raspberry-Pi

Fetches the WebRTC source code using depot_tools and installs the necessary sysroot for ARM architecture.

```bash
fetch --no-history webrtc
src/build/linux/sysroot_scripts/install-sysroot.py --arch=arm
```

--------------------------------

### Install depot_tools

Source: https://github.com/mpromonet/webrtc-streamer/wiki/Cross-compile-WebRTC-for-Raspberry-Pi

Clones the depot_tools repository and adds it to the system's PATH for managing WebRTC dependencies.

```bash
sudo git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git /opt/depot_tools
echo "export PATH=/opt/depot_tools:$PATH" | sudo tee /etc/profile.d/depot_tools.sh

source /etc/profile
```

--------------------------------

### Install Chromium Depot Tools

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Clone the depot_tools repository and add it to the PATH. This is the first step in building WebRTC Streamer.

```sh
pushd ..
git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git
export PATH=$PATH:`realpath depot_tools`
popd
```

--------------------------------

### Display a grid of WebRTC streams

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

This example demonstrates how to arrange multiple WebRTC streams in a grid layout, specified by the 'layout' parameter.

```html
?layout=2x4
```

--------------------------------

### Start Embedded STUN/TURN Server

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Start the webrtc-streamer with an embedded STUN and TURN server. The public IP is dynamically fetched using curl.

```sh
./webrtc-streamer --stun-server=0.0.0.0:3478 --stun=$(curl -s ifconfig.me):3478
```

```sh
./webrtc-streamer --stun=- --turn-server=0.0.0.0:3478 -tturn:turn@$(curl -s ifconfig.me):3478
```

```sh
./webrtc-streamer --stun-server=0.0.0.0:3478 --stun=$(curl -s ifconfig.me):3478 --turn-server=0.0.0.0:3479 --turn=turn:turn@$(curl -s ifconfig.me):3479
```

--------------------------------

### Run Basic Docker Image

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Start the webrtc-streamer application using its Docker image. This command exposes the application on port 8000.

```sh
docker run -p 8000:8000 -it mpromonet/webrtc-streamer
```

--------------------------------

### CPack Packaging Configuration

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Configures installation rules and CPack generator settings for creating distributable packages.

```cmake
install (TARGETS ${CMAKE_PROJECT_NAME} RUNTIME DESTINATION bin)
install (DIRECTORY html DESTINATION share/${CMAKE_PROJECT_NAME})
install (FILES config.json DESTINATION share/${CMAKE_PROJECT_NAME})
SET(CPACK_GENERATOR "TGZ")
SET(CPACK_SYSTEM_NAME ${CMAKE_SYSTEM_NAME}-${CMAKE_SYSTEM_PROCESSOR}-${CMAKE_BUILD_TYPE})
SET(CPACK_PACKAGE_CONTACT "michel.promonet@free.fr")
if(PROJECTVERSION)
    SET(CPACK_PACKAGE_VERSION "${PROJECTVERSION}")
endif()
INCLUDE(CPack)
```

--------------------------------

### Install Raspberry Pi GCC Cross-Compiler

Source: https://github.com/mpromonet/webrtc-streamer/wiki/Cross-compile-WebRTC-for-Raspberry-Pi

Downloads and extracts a specific GCC cross-compiler toolchain for Raspberry Pi 2/3 (Buster) and adds its bin directory to the PATH.

```bash
wget -qO- https://sourceforge.net/projects/raspberry-pi-cross-compilers/files/Raspberry%20Pi%20GCC%20Cross-Compiler%20Toolchains/Buster/GCC%209.3.0/Raspberry%20Pi%202%2C%203/cross-gcc-9.3.0-pi_2-3.tar.gz  | sudo tar xz -C /opt 
echo "export PATH=$(ls -d /opt/cross-pi-gcc-*/bin):$PATH" | sudo tee /etc/profile.d/arm_tools.sh

source /etc/profile
```

--------------------------------

### Minimal WHEP Player Example

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

A minimal HTML page using the @eyevinn/whep-video-component to display a WebRTC stream via the WHEP protocol. It configures the video source to stream from the webrtc-streamer API.

```html
<html>
<head>
    <script src="https://unpkg.com/@eyevinn/whep-video-component@latest/dist/whep-video.component.js"></script>
</head>
<body>
    <whep-video id="video" muted autoplay></whep-video>
    <script>
        video.setAttribute('src', `${location.origin}/api/whep?url=Asahi&options=rtptransport%3dtcp%26timeout%3d60`);
    </script>
</body>
</html>
```

--------------------------------

### Configure NAT Rules with upnpc

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Use upnpc to configure NAT rules for port forwarding. This example forwards ports for HTTP, STUN, and TURN.

```sh
upnpc -r 8000 tcp 3478 tcp 3478 udp
```

--------------------------------

### WebRTC Build Arguments Configuration

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Sets various build arguments for the WebRTC library, such as disabling tests, examples, and enabling/disabling specific features like H.264/H.265 support based on build type and branding.

```cmake
set (WEBRTCARGS rtc_include_tests=false
rtc_enable_protobuf=false
rtc_build_examples=false
treat_warnings_as_errors=false
enable_js_protobuf=false
use_glib=false
)
set (WEBRTCARGS libyuv_use_sme=false
${WEBRTCARGS})
if (CMAKE_CXX_COMPILER_ID STREQUAL "Clang" AND NOT WIN32)
  set (WEBRTCARGS use_custom_libcxx=true
${WEBRTCARGS})
else()
  set (WEBRTCARGS use_custom_libcxx=false
${WEBRTCARGS})
endif()
# debug/release
if(CMAKE_BUILD_TYPE STREQUAL "Release")
  set (WEBRTCARGS is_debug=false
${WEBRTCARGS})
else()
  set (WEBRTCARGS is_debug=true
${WEBRTCARGS})
endif()

# enable H264 support
if (WEBRTCCHROMEBRANDED STREQUAL "ON")
  set (WEBRTCARGS is_chrome_branded=true
${WEBRTCARGS})
else()
  set (WEBRTCARGS rtc_use_h264=true
rtc_use_h265=true
${WEBRTCARGS})
endif()
```

--------------------------------

### Get Ice Candidates

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Retrieves the list of ICE candidates for the communication.

```APIDOC
## GET /api/getIceCandidate

### Description
Retrieves the list of ICE candidates for the communication.

### Method
GET

### Endpoint
/api/getIceCandidate
```

--------------------------------

### Get Ice Configuration

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Retrieves the IceServers configuration used by the streamer side.

```APIDOC
## GET /api/getIceServers

### Description
Retrieves the IceServers configuration used by the streamer side.

### Method
GET

### Endpoint
/api/getIceServers
```

--------------------------------

### HTML Page with WebComponent

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

An example HTML page using the webrtc-streamer Web Component. This provides a declarative way to embed WebRTC streams directly in HTML.

```html
<html>
<head>
	<script type="module" src="webrtc-streamer-element.js"></script>
</head>
<body>
	<webrtc-streamer url="rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov"></webrtc-streamer>
</body>
</html>
```

--------------------------------

### Determine Project Version using Git

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

This snippet dynamically determines the project version by executing Git commands to get tags, commit hashes, and dirty status. It also appends version information for submodules like civetweb, webrtc, and live555helper. Use this to embed version information directly into the build.

```cmake
if(GIT_FOUND)
    EXECUTE_PROCESS(COMMAND ${GIT_EXECUTABLE} submodule update --init)
    EXECUTE_PROCESS(COMMAND ${GIT_EXECUTABLE} describe --tags --always --dirty OUTPUT_VARIABLE PROJECTVERSION OUTPUT_STRIP_TRAILING_WHITESPACE)
    set (VERSION "${PROJECTVERSION}/${CMAKE_SYSTEM_NAME}-${CMAKE_SYSTEM_PROCESSOR}")

    EXECUTE_PROCESS(COMMAND ${GIT_EXECUTABLE} -C civetweb describe --tags --always --dirty OUTPUT_VARIABLE CIVETVERSION OUTPUT_STRIP_TRAILING_WHITESPACE)
    set (VERSION "${VERSION} civetweb@${CIVETVERSION}")

    EXECUTE_PROCESS(COMMAND ${GIT_EXECUTABLE} -C ${WEBRTCROOT}/src describe --tags --always --dirty OUTPUT_VARIABLE WEBRTCVERSION OUTPUT_STRIP_TRAILING_WHITESPACE)
    set (VERSION "${VERSION} webrtc@${WEBRTCVERSION}")

    EXECUTE_PROCESS(COMMAND ${GIT_EXECUTABLE} -C live555helper describe --tags --always --dirty OUTPUT_VARIABLE LIVEVERSION OUTPUT_STRIP_TRAILING_WHITESPACE)
    set (VERSION "${VERSION} live555helper@${LIVEVERSION}")
endif()
add_definitions(-DVERSION=\"${VERSION}\")
MESSAGE("VERSION = ${VERSION}")
add_definitions(-DWEBRTCSTREAMERRESSOURCE=\"${WEBRTCSTREAMERRESSOURCE}\")
MESSAGE("WEBRTCSTREAMERRESSOURCE = ${WEBRTCSTREAMERRESSOURCE}")
```

--------------------------------

### Download WebRTC Source

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Create a directory for WebRTC source code and fetch the repository using depot_tools.

```sh
mkdir ../webrtc
pushd ../webrtc
fetch webrtc 
popd
```

--------------------------------

### View Docker Image Help

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Run the webrtc-streamer Docker image with the --help flag to view all available commands and options.

```sh
docker run -p 8000:8000 -it mpromonet/webrtc-streamer --help
```

--------------------------------

### Initialize JanusVideoRoom

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Instantiate the JanusVideoRoom class with Janus and WebRTC streamer URLs.

```js
var janus = new JanusVideoRoom(<janus url>, <webrtc-streamer url>)
```

--------------------------------

### Initialize XMPPVideoRoom for Jitsi

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Instantiate the XMPPVideoRoom class with the XMPP server URL and WebRTC streamer URL.

```js
var xmpp = new XMPPVideoRoom(<xmpp server url>, <webrtc-streamer url>)
```

--------------------------------

### Configure WebRTC Build with GN

Source: https://github.com/mpromonet/webrtc-streamer/wiki/Cross-compile-WebRTC-for-Raspberry-Pi

Generates build files using GN for an ARM release build, enabling H.264, disabling tests, and configuring other build options.

```bash
pushd src 
gn gen arm/out/Release --args='is_debug=false rtc_use_h264=true ffmpeg_branding="Chrome" is_clang=false target_cpu="arm" treat_warnings_as_errors=false rtc_include_tests=false rtc_enable_protobuf=false use_custom_libcxx=false use_ozone=true rtc_include_pulse_audio=false rtc_build_examples=false'
popd
```

--------------------------------

### Display webrtc-streamer help message

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Use this command to display all available options and their descriptions for the webrtc-streamer tool.

```bash
./webrtc-streamer [OPTION...] [urls...]
```

--------------------------------

### Linux Desktop Capture and X11 Libraries

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Conditionally enables desktop capture if the ninja build files exist and links necessary X11 libraries.

```cmake
if (EXISTS ${WEBRTCROOT}/src/out/${CMAKE_BUILD_TYPE}/obj/modules/desktop_capture/desktop_capture.ninja)
    target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE USE_X11)
    find_package(X11)
    target_link_libraries(${CMAKE_PROJECT_NAME} ${X11_LIBRARIES} ${X11_Xdamage_LIB} ${X11_Xfixes_LIB} ${X11_Xcomposite_LIB} ${X11_Xrandr_LIB} ${X11_Xtst_LIB})
endif()
```

--------------------------------

### Build WebRTC Streamer

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Configure the build using CMake, specifying the clang compiler, and then compile the project. Ensure the LLVM bin directory is in the PATH.

```sh
export PATH=$PATH:`realpath ../webrtc/src/third_party/llvm-build/Release+Asserts/bin`
cmake -DCMAKE_C_COMPILER=clang . 
make
```

--------------------------------

### Civetweb Library Configuration

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Adds Civetweb as a static library, sets its compile definitions for OpenSSL and WebSockets, links it to the main project, and sets its include directories.

```cmake
add_library (civetweb STATIC civetweb/src/civetweb.c civetweb/src/CivetServer.cpp)
target_compile_definitions(civetweb PUBLIC OPENSSL_API_3_0 USE_WEBSOCKET)
target_link_libraries (${CMAKE_PROJECT_NAME} civetweb)
target_include_directories(civetweb PUBLIC civetweb/include)
```

--------------------------------

### Configure live555helper

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Adds the live555helper subdirectory and links it along with boringssl. Defines HAVE_LIVE555.

```cmake
add_subdirectory(live555helper EXCLUDE_FROM_ALL)
target_include_directories(liblive555helper PRIVATE ${WEBRTCROOT}/src/third_party/boringssl/src/include)
target_link_libraries (${CMAKE_PROJECT_NAME} liblive555helper ${WEBRTCOBJS}/third_party/boringssl/${CMAKE_STATIC_LIBRARY_PREFIX}boringssl${CMAKE_STATIC_LIBRARY_SUFFIX}) 
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE HAVE_LIVE555)
```

--------------------------------

### Build WebRTC with Ninja

Source: https://github.com/mpromonet/webrtc-streamer/wiki/Cross-compile-WebRTC-for-Raspberry-Pi

Compiles the WebRTC project using Ninja for the previously configured ARM release build.

```bash
pushd src 
ninja -C arm/out/Release
popd
```

--------------------------------

### Add Prometheus-cpp Core

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Includes the Prometheus-cpp core library and links it to the project. Sets up include directories.

```cmake
include(GenerateExportHeader)
include(GNUInstallDirs)
set(PROJECT_NAME prometheus-cpp)
add_subdirectory(prometheus-cpp/core EXCLUDE_FROM_ALL)
target_link_libraries (${CMAKE_PROJECT_NAME} core)
target_include_directories(${CMAKE_PROJECT_NAME} PRIVATE prometheus-cpp/core/include)
```

--------------------------------

### WebRTC Include Directories

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Sets the include directories for WebRTC and its dependencies. Ensure WEBRTCROOT is correctly defined.

```cmake
target_include_directories(${CMAKE_PROJECT_NAME} PRIVATE ${WEBRTCROOT}/src ${WEBRTCROOT}/src/third_party/abseil-cpp ${WEBRTCROOT}/src/third_party/jsoncpp/source/include  ${WEBRTCROOT}/src/third_party/jsoncpp/generated ${WEBRTCROOT}/src/third_party/libyuv/include)
```

--------------------------------

### Configure sound support based on OS

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Sets WEBRTCARGS based on the operating system to include or exclude specific audio device support.

```cmake
if (APPLE)
  set (WEBRTCARGS rtc_include_internal_audio_device=true
${WEBRTCARGS})
  set (WEBRTCARGS rtc_include_pulse_audio=true
${WEBRTCARGS})
elseif (WIN32)
  set (WEBRTCARGS rtc_include_internal_audio_device=true
${WEBRTCARGS})
elseif (NOT DEFINED CMAKE_SYSROOT)
  if (NOT PulseAudio_FOUND)
    set (WEBRTCARGS rtc_include_pulse_audio=false
${WEBRTCARGS})
  endif()
  if (NOT ALSA_FOUND)
    set (WEBRTCARGS rtc_include_internal_audio_device=false
${WEBRTCARGS})
  endif()
endif()
```

--------------------------------

### Linux Civetweb and ALSA Libraries

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Links the dl library for Civetweb and conditionally defines HAVE_SOUND if ALSA is found.

```cmake
target_link_libraries(${CMAKE_PROJECT_NAME} dl)
if (ALSA_FOUND)
    target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE HAVE_SOUND)
endif()
```

--------------------------------

### Define Project Executable and Link Libraries

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

This snippet defines the main executable for the project using `add_executable` and `GLOB_RECURSE` to find source files. It then links necessary libraries, including custom libc++ and libc++abi if found, and the Threads library. Ensure that the paths to WEBRTCOBJS and source files are correctly configured.

```cmake
FILE(GLOB_RECURSE WEBRTJSONCPPCOBJS ${WEBRTCOBJS}/third_party/jsoncpp/jsoncpp/*${CMAKE_C_OUTPUT_EXTENSION} ${WEBRTCOBJS}/rtc_base/rtc_json/*${CMAKE_C_OUTPUT_EXTENSION})
FILE(GLOB_RECURSE WEBRTP2POBJ ${WEBRTCOBJS}/p2p/p2p_server_utils/*${CMAKE_C_OUTPUT_EXTENSION})
SET (WEBRTCEXTRAOBJS ${WEBRTJSONCPPCOBJS} ${WEBRTP2POBJ} ${WEBRTCCOMOBJ})
FILE(GLOB SOURCE src/*.cpp)
add_executable (${CMAKE_PROJECT_NAME} ${SOURCE} ${WEBRTCEXTRAOBJS})
target_include_directories(${CMAKE_PROJECT_NAME} PRIVATE inc)
if (EXISTS ${WEBRTCOBJS}/buildtools/third_party/libc++/${CMAKE_STATIC_LIBRARY_PREFIX}c++${CMAKE_STATIC_LIBRARY_SUFFIX})
  message(STATUS "Found libc++ library at ${WEBRTCOBJS}/buildtools/third_party/libc++/${CMAKE_STATIC_LIBRARY_PREFIX}c++${CMAKE_STATIC_LIBRARY_SUFFIX}")
  target_link_libraries(${CMAKE_PROJECT_NAME} "${WEBRTCOBJS}/buildtools/third_party/libc++/${CMAKE_STATIC_LIBRARY_PREFIX}c++${CMAKE_STATIC_LIBRARY_SUFFIX}")
endif()
if (EXISTS ${WEBRTCOBJS}/buildtools/third_party/libc++abi/${CMAKE_STATIC_LIBRARY_PREFIX}c++abi${CMAKE_STATIC_LIBRARY_SUFFIX})
  message(STATUS "Found libc++abi library at ${WEBRTCOBJS}/buildtools/third_party/libc++abi/${CMAKE_STATIC_LIBRARY_PREFIX}c++abi${CMAKE_STATIC_LIBRARY_SUFFIX}")
  target_link_libraries(${CMAKE_PROJECT_NAME} "${WEBRTCOBJS}/buildtools/third_party/libc++abi/${CMAKE_STATIC_LIBRARY_PREFIX}c++abi${CMAKE_STATIC_LIBRARY_SUFFIX}")
endif()

# cxxopts
target_include_directories(${CMAKE_PROJECT_NAME} PRIVATE cxxopts/include)
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE CXXOPTS_NO_RTTI)

# thread
if (NOT WIN32)
  set(CMAKE_THREAD_LIBS_INIT "-lpthread")
endif()
find_package(Threads REQUIRED)
target_link_libraries(${CMAKE_PROJECT_NAME} Threads::Threads)
```

--------------------------------

### Run Docker Image with Config File

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Mount a custom config.json file into the webrtc-streamer Docker container. This allows for custom configuration of the streamer.

```sh
docker run --name webrtc -d -p 8000:8000 -v $PWD/config.json:/usr/local/share/webrtc-streamer/config.json mpromonet/webrtc-streamer
```

--------------------------------

### Configure screen capture support

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Disables X11 and PipeWire for screen capture if GTK3 is not found or WEBRTCDESKTOPCAPTURE is OFF.

```cmake
find_package(PkgConfig QUIET)
pkg_check_modules(GTK3 QUIET gtk+-3.0)
MESSAGE("GTK3_FOUND = ${GTK3_FOUND}")
if(NOT GTK3_FOUND OR (WEBRTCDESKTOPCAPTURE STREQUAL "OFF"))
  set (WEBRTCARGS rtc_use_x11=false
rtc_use_pipewire=false
${WEBRTCARGS})
endif()
```

--------------------------------

### Publish WebRTC Streams to Jitsi Video Room

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Join a Jitsi Video Room with the specified room name, stream URL, and display name using the XMPPVideoRoom interface. Requires several Strophe and Jingle libraries.

```html
<html>
<head>
<script src="libs/strophe.min.js" ></script>
<script src="libs/strophe.muc.min.js" ></script>
<script src="libs/strophe.disco.min.js" ></script>
<script src="libs/strophe.jingle.sdp.js"></script>
<script src="libs/jquery-3.5.1.min.js"></script>
<script src="xmppvideoroom.js" ></script>
<script>
	var xmpp = new XMPPVideoRoom("meet.jit.si", null);
	muxpp.join("testroom", "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov","Bunny");
</script>
</head>
</html>
```

--------------------------------

### Write build arguments to file

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Writes the accumulated WEBRTCARGS to the args.gn file in the build output directory.

```cmake
FILE(WRITE ${WEBRTCROOT}/src/out/${CMAKE_BUILD_TYPE}/args.gn ${WEBRTCARGS})
```

--------------------------------

### Configure compilation mode for ARM processors

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Sets WEBRTCARGS for different ARM processor architectures, including target CPU, version, float ABI, and libaom enablement.

```cmake
if(CMAKE_SYSTEM_PROCESSOR MATCHES "armv6.*")
  set (WEBRTCARGS target_cpu="arm"
arm_version=6
arm_float_abi="hard"
enable_libaom=false
${WEBRTCARGS})
elseif(CMAKE_SYSTEM_PROCESSOR MATCHES "armv.*")
  set (WEBRTCARGS target_cpu="arm"
${WEBRTCARGS})
elseif(CMAKE_SYSTEM_PROCESSOR MATCHES "arm64")
  set (WEBRTCARGS target_cpu="arm64"
${WEBRTCARGS})
endif()
```

--------------------------------

### Initiate Communication Asking to be Called

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Creates an offer to initiate communication, allowing the other party to call back.

```APIDOC
## POST /api/createOffer

### Description
Creates an offer to initiate communication, allowing the other party to call back.

### Method
POST

### Endpoint
/api/createOffer
```

--------------------------------

### Publish WebRTC Streams to Janus Video Room (NodeJS)

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Join a Janus Video Room using the JanusVideoRoom module in NodeJS. Requires 'then-request' and the local 'janusvideoroom.js' module.

```js
global.request = require("then-request");
var JanusVideoRoom = require("./html/janusvideoroom.js");
var janus = new JanusVideoRoom(
  "http://192.168.0.15:8088/janus",
  "http://192.168.0.15:8000",
);
janus.join(1234, "videocap://0", "video");
```

--------------------------------

### Linux libv4l2cpp Library

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Adds the libv4l2cpp library as a static library, sets its include directories, and links it to the main project. Also defines HAVE_V4L2.

```cmake
aux_source_directory(libv4l2cpp/src LIBSRC_FILES)
add_library(libv4l2cpp STATIC ${LIBSRC_FILES})
target_include_directories(libv4l2cpp PUBLIC libv4l2cpp/inc)
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE HAVE_V4L2)
target_link_libraries (${CMAKE_PROJECT_NAME} libv4l2cpp)
```

--------------------------------

### Generate build files with gn

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Executes the 'gn gen' command to generate build files for the specified build type. Sets the shell command based on the operating system.

```cmake
if (WIN32)
  SET (SHELLCOMMAND cmd /c )
endif()
EXECUTE_PROCESS(WORKING_DIRECTORY ${WEBRTCROOT}/src/out/${CMAKE_BUILD_TYPE} COMMAND ${SHELLCOMMAND} gn gen . RESULT_VARIABLE NINJA_RESULT)
```

--------------------------------

### Build webrtc with ninja

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Compiles the webrtc project using ninja with a specified list of targets. Checks for build failures and reports an error if the build fails.

```cmake
SET(NINJA_TARGET webrtc rtc_json builtin_video_decoder_factory builtin_video_encoder_factory p2p_server_utils task_queue default_task_queue_factory field_trials rtc_tools)
EXECUTE_PROCESS(WORKING_DIRECTORY ${WEBRTCROOT}/src/out/${CMAKE_BUILD_TYPE} COMMAND ${SHELLCOMMAND} ninja ${NINJA_TARGET} RESULT_VARIABLE NINJA_RESULT)
if(NOT NINJA_RESULT EQUAL 0)
  message(FATAL_ERROR "webrtc build failed with exit code ${NINJA_RESULT}")
endif()
```

--------------------------------

### Initiate Communication as Caller

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Sends an offer and receives an answer to initiate communication as the caller.

```APIDOC
## POST /api/call

### Description
Sends an offer and receives an answer to initiate communication as the caller.

### Method
POST

### Endpoint
/api/call
```

--------------------------------

### Copying FFmpeg Configuration Files

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Copies generated FFmpeg configuration header files to specific locations within the WebRTC build directory. This is part of the FFmpeg integration process for ARMv6 builds.

```cmake
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/config.h DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/config_components.h DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavutil/avconfig.h DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavutil/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavfilter/filter_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavfilter/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavcodec/codec_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavcodec/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavcodec/parser_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavcodec/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavcodec/bsf_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavcodec/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavformat/demuxer_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavformat/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavformat/muxer_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavformat/)
file(COPY ${WEBRTCROOT}/src/third_party/ffmpeg/libavformat/protocol_list.c DESTINATION ${WEBRTCROOT}/src/third_party/ffmpeg/chromium/config/Chrome/linux/arm/libavformat/)
```

--------------------------------

### HTML Page with WebRTC Stream

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

A sample HTML page demonstrating how to use the WebRtcStreamer JavaScript class to display a video stream. It connects to a local webrtc-streamer instance and plays an RTSP stream.

```html
<html>
<head>
<script src="libs/adapter.min.js" ></script>
<script src="webrtcstreamer.js" ></script>
<script>
	var webRtcServer      = null;
	window.onload         = function() { 
		webRtcServer      = new WebRtcStreamer("video",location.protocol+"//"+location.hostname+":8000");
		webRtcServer.connect("rtsp://196.21.92.82/axis-media/media.amp","", "rtptransport=tcp&timeout=60");
	}
	window.onbeforeunload = function() { webRtcServer.disconnect(); }
</script>
</head>
<body> 
	<video id="video" muted playsinline />
</body>
</html>
```

--------------------------------

### WebRTC Compile Definitions

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Adds compile definitions for WebRTC, including HAVE_JPEG.

```cmake
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE HAVE_JPEG)
```

--------------------------------

### Publish WebRTC Streams to Janus Video Room (Browser)

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Join a Janus Video Room with specified room ID, stream URL, and display name using the JanusVideoRoom interface in a browser.

```html
<html>
<head>
<script src="janusvideoroom.js" ></script>
<script>
	var janus = new JanusVideoRoom("https://janus.conf.meetecho.com/janus", null);
	janus.join(1234, "rtsp://pi2.local:8554/unicast","pi2");
	janus.join(1234, "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov","media");
</script>
</head>
</html>
```

--------------------------------

### Configure RTMP Support

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Finds and configures RTMP library support. Links the RTMP library if found.

```cmake
find_package(PkgConfig QUIET)
pkg_check_modules(RTMP QUIET librtmp)
MESSAGE("RTMP_FOUND = ${RTMP_FOUND} RTMP_INCLUDE_DIRS=${RTMP_INCLUDE_DIRS} RTMP_LIBRARY_DIRS=${RTMP_LIBRARY_DIRS} RTMP_LIBRARIES=${RTMP_LIBRARIES}")
find_library(RTMP_LIB NAMES rtmp)
MESSAGE("RTMP_LIB=${RTMP_LIB}")
if (RTMP_FOUND AND RTMP_LIB_FOUND)
  target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE HAVE_RTMP)
  target_include_directories(${CMAKE_PROJECT_NAME} PUBLIC ${RTMP_INCLUDE_DIRS})
  target_link_directories(${CMAKE_PROJECT_NAME} PUBLIC ${RTMP_LIBRARY_DIRS})
  target_link_libraries (${CMAKE_PROJECT_NAME} ${RTMP_LIB})
endif()
```

--------------------------------

### Windows WebRTC Definitions and Libraries

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Configures WebRTC specific compile definitions and links necessary Windows libraries for multimedia and networking.

```cmake
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE WEBRTC_WIN EHsc NOMINMAX __PRETTY_FUNCTION__=__FUNCTION__)
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE USE_X11 HAVE_SOUND)
target_link_libraries (${CMAKE_PROJECT_NAME} secur32 dmoguids wmcodecdspuuid strmiids msdmo winmm dxgi d3d11 iphlpapi dwmapi shcore)
```

--------------------------------

### macOS WebRTC Definitions and Libraries

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Configures WebRTC specific compile definitions and finds/links macOS frameworks for audio, video, and application services.

```cmake
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE WEBRTC_MAC WEBRTC_POSIX)
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE HAVE_SOUND)
add_definitions(-fno-rtti)

find_library(CORE_FOUNDATION Foundation)
find_library(APPLICATION_SERVICES ApplicationServices)
find_library(CORE_SERVICES CoreServices)
find_library(CORE_AUDIO CoreAudio)
find_library(AUDIO_TOOLBOX AudioToolBox)
find_library(IO_SURFACE IOSurface)
find_library(APP_KIT AppKit)


target_link_libraries (${CMAKE_PROJECT_NAME} ${CORE_FOUNDATION} ${APPLICATION_SERVICES} ${CORE_SERVICES}
                       ${CORE_AUDIO} ${AUDIO_TOOLBOX} ${IO_SURFACE} ${APP_KIT})
```

--------------------------------

### Windows Live555 Definitions

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Adds definitions for Live555 on Windows to disable iterator debug checking and enable Winsock.

```cmake
add_definitions(-D_HAS_ITERATOR_DEBUGGING=0 -D_WINSOCKAPI_)
```

--------------------------------

### macOS Live555 Definitions

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Adds definitions for Live555 on macOS to enable locale support.

```cmake
add_definitions(-DNEED_XLOCALE_H=1)
```

--------------------------------

### Run Docker Image with Extra Host

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Run the webrtc-streamer Docker container and add an extra host entry for host.docker.internal. This is useful for accessing host services from within the container.

```sh
docker run --name webrtc -d -p 8000:8000 --add-host host.docker.internal:host-gateway mpromonet/webrtc-streamer -e host.docker.internal
```

--------------------------------

### Run Docker Image with Host Network

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Run the webrtc-streamer Docker container using the host's network stack. This bypasses Docker's network isolation.

```sh
docker run --name webrtc -d --net host mpromonet/webrtc-streamer
```

--------------------------------

### Run Docker Image with V4L2 Device

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Expose V4L2 devices from the host machine to the webrtc-streamer Docker container. This allows the container to access host video devices.

```sh
docker run --device=/dev/video0 -p 8000:8000 -it mpromonet/webrtc-streamer
```

--------------------------------

### FFmpeg Configuration for ARMv6

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Configures FFmpeg with specific options for ARMv6 architecture, including enabling certain codecs and demuxers, and setting cross-compilation flags. This snippet is executed only if the system processor matches 'armv6'.

```cmake
if (CMAKE_SYSTEM_PROCESSOR MATCHES "armv6.*" )
  set (CONFIGURE_ARGS --disable-everything --disable-all --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-static --enable-avcodec --enable-avformat --enable-avutil --enable-static --enable-libopus --disable-debug --disable-bzlib --disable-iconv --disable-network --disable-schannel --disable-sdl2 --disable-symver --disable-xlib --disable-zlib --disable-securetransport --disable-faan --disable-alsa --disable-autodetect --disable-error-resilience --enable-decoder=vorbis,libopus,flac,pcm_u8,pcm_s16le,pcm_s24le,pcm_s32le,pcm_f32le,mp3,pcm_s16be,pcm_s24be,pcm_mulaw,pcm_alaw --enable-demuxer=ogg,matroska,wav,flac,mp3,mov --enable-parser=opus,vorbis,flac,mpegaudio,vp9 --extra-cflags=-I${WEBRTCROOT}/src/third_party/opus/src/include --disable-linux-perf --x86asmexe=nasm --optflags="-O2" --arch=arm --enable-armv6 --disable-armv6t2 --enable-vfp --disable-thumb --extra-cflags=-march=armv6 --enable-cross-compile --target-os=linux --extra-cflags=--target=arm-linux-gnueabihf --extra-ldflags=--target=arm-linux-gnueabihf --sysroot=${WEBRTCROOT}/src/build/linux/debian_bullseye_armhf-sysroot --extra-cflags=-mtune=cortex-a8 --extra-cflags=-mfloat-abi=hard --extra-cflags=-O2 --disable-neon --extra-cflags=-mfpu=vfpv3-d16 --enable-pic --cc=clang --cxx=clang++ --ld=clang --extra-ldflags=-fuse-ld=lld --enable-decoder=aac,h264 --enable-demuxer=aac --enable-parser=aac,h264)
MESSAGE("CONFIGURE_ARGS = ${CONFIGURE_ARGS}")
EXECUTE_PROCESS(WORKING_DIRECTORY ${WEBRTCROOT}/src/third_party/ffmpeg COMMAND ./configure ${CONFIGURE_ARGS})
```

--------------------------------

### Initiate Communication using WHEP

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Initiates communication using the WHEP protocol, similar to the /api/call endpoint.

```APIDOC
## POST /api/whep

### Description
Initiates communication using the WHEP protocol, similar to the /api/call endpoint.

### Method
POST

### Endpoint
/api/whep
```

--------------------------------

### Linux WebRTC Definitions and RTTI

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Configures WebRTC specific compile definitions for POSIX systems and disables RTTI. Includes ARM-specific flags for armv6 processors.

```cmake
target_compile_definitions(${CMAKE_PROJECT_NAME} PRIVATE WEBRTC_POSIX)
add_definitions(-fno-rtti -fpermissive)
if (CMAKE_SYSTEM_PROCESSOR MATCHES "armv6.*")
    add_definitions(-marm -march=armv6 -mfpu=vfp -mfloat-abi=hard)
endif()
```

--------------------------------

### Initialize WebRtcStreamer in JavaScript

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Instantiate the WebRtcStreamer class in JavaScript to embed a WebRTC stream in an HTML page. Requires the video tag ID and the webrtc-streamer URL.

```javascript
var webRtcServer = new WebRtcStreamer(<video tag>, <webrtc-streamer url>);
```

--------------------------------

### Configure RTX stream config

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Modifies the rtp_config.cc file to configure RTX stream settings.

```cmake
string(REGEX REPLACE "stream_config.rtx.emplace\(\)" "stream_config.rtx.emplace(RtpStreamConfig::Rtx())" filecontent "${filecontent}")
file(WRITE ${WEBRTCROOT}/src/call/rtp_config.cc "${filecontent}")
```

--------------------------------

### Configure Clang for libc++ on non-Windows

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

This configuration block is executed when the compiler is Clang and the system is not Windows. It sets flags to use libc++ and includes necessary system headers for libc++ and libc++abi. Ensure that the libc++ and libc++abi paths are correctly set relative to your project structure.

```cmake
if (CMAKE_CXX_COMPILER_ID STREQUAL "Clang")
  if (NOT WIN32)
    message(STATUS "Using Clang: enabling libc++")

    set (CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -nostdinc++")
    add_compile_definitions(_LIBCPP_HARDENING_MODE=_LIBCPP_HARDENING_MODE_EXTENSIVE)
    add_compile_options(-I${WEBRTCROOT}/src/buildtools/third_party/libc++)
    if (EXISTS ${WEBRTCBUILD}/gen/third_party/libc++/src/include)
      add_compile_options(-isystem${WEBRTCBUILD}/gen/third_party/libc++/src/include)
    else()
      add_compile_options(-isystem${WEBRTCROOT}/src/third_party/libc++/src/include)
    endif()
    add_compile_options(-isystem${WEBRTCROOT}/src/third_party/libc++abi/src/include)
    add_compile_options(-Wno-nullability-completeness)
  endif()

  message(STATUS "Compiler is Clang CMAKE_LIBRARY_ARCHITECTURE:${CMAKE_LIBRARY_ARCHITECTURE} CMAKE_SYSTEM_PROCESSOR:${CMAKE_SYSTEM_PROCESSOR}")
  if(CMAKE_SYSTEM_PROCESSOR MATCHES "armv.*")
    if (DEFINED CMAKE_SYSROOT)
      add_compile_options(-isystem${CMAKE_SYSROOT}/usr/include/arm-linux-gnueabihf)
    endif()
    add_compile_options(--target=arm-linux-gnueabihf)
    add_link_options(--target=arm-linux-gnueabihf)
  elseif(CMAKE_SYSTEM_PROCESSOR MATCHES "arm64")
    if (NOT APPLE)
      if (DEFINED CMAKE_SYSROOT)
        add_compile_options(-isystem${CMAKE_SYSROOT}/usr/include/aarch64-linux-gnu)
      endif()
      add_compile_options(--target=aarch64-linux-gnu)
      add_link_options(--target=aarch64-linux-gnu)
    else ()
      add_compile_options(--target=aarch64-apple-macos)
      add_link_options(--target=aarch64-apple-macos)
    endif()
  endif()
endif()
```

--------------------------------

### WebRTC Libraries Linking

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Links essential WebRTC libraries to the project. The list includes various API and media components.

```cmake
set(WEBRTC_LIBS_INFO
    "${WEBRTCOBJS}/api/${CMAKE_STATIC_LIBRARY_PREFIX}field_trials${CMAKE_STATIC_LIBRARY_SUFFIX}"
    "${WEBRTCOBJS}/api/video_codecs/${CMAKE_STATIC_LIBRARY_PREFIX}rtc_software_fallback_wrappers${CMAKE_STATIC_LIBRARY_SUFFIX}"
    "${WEBRTCOBJS}/api/video_codecs/${CMAKE_STATIC_LIBRARY_PREFIX}builtin_video_encoder_factory${CMAKE_STATIC_LIBRARY_SUFFIX}"
    "${WEBRTCOBJS}/api/video_codecs/${CMAKE_STATIC_LIBRARY_PREFIX}builtin_video_decoder_factory${CMAKE_STATIC_LIBRARY_SUFFIX}"
    "${WEBRTCOBJS}/media/${CMAKE_STATIC_LIBRARY_PREFIX}rtc_internal_video_codecs${CMAKE_STATIC_LIBRARY_SUFFIX}"
    "${WEBRTCOBJS}/media/${CMAKE_STATIC_LIBRARY_PREFIX}rtc_simulcast_encoder_adapter${CMAKE_STATIC_LIBRARY_SUFFIX}"
    "${WEBRTCOBJS}/${CMAKE_STATIC_LIBRARY_PREFIX}webrtc${CMAKE_STATIC_LIBRARY_SUFFIX}"
)
foreach(LIB ${WEBRTC_LIBS_INFO})
    target_link_libraries (${CMAKE_PROJECT_NAME} ${LIB})
endforeach()
```

--------------------------------

### Run Docker Image with RTSP URL

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Register an RTSP URL with the webrtc-streamer Docker container. The container will stream from the specified RTSP source.

```sh
docker run --name webrtc -d -p 8000:8000 -it mpromonet/webrtc-streamer -n raspicam -u rtsp://pi2.local:8554/unicast
```

--------------------------------

### Access a named WebRTC stream

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Connect to a WebRTC stream using a predefined name 'Bunny'. This assumes the stream has been previously configured or registered with that name.

```html
webrtcstreamer.html?Bunny
```

--------------------------------

### macOS Linker Flags

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Sets the linker flags to use lld and specifies the path to the LLVM bin directory for macOS builds.

```cmake
set(CMAKE_EXE_LINKER_FLAGS "${CMAKE_EXE_LINKER_FLAGS} -fuse-ld=lld -B${WEBRTCROOT}/src/third_party/llvm-build/Release+Asserts/bin")
```

--------------------------------

### Add Ice Candidate

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Adds a new ICE candidate for the communication.

```APIDOC
## POST /api/addIceCandidate

### Description
Adds a new ICE candidate for the communication.

### Method
POST

### Endpoint
/api/addIceCandidate
```

--------------------------------

### Stream RTSP to WebRTC

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/README.md

Access a live RTSP stream and make it available via WebRTC. The stream name 'Bunny' is used as an alias.

```html
webrtcstreamer.html?rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov
```

--------------------------------

### Configure sysroot usage

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Sets the 'use_sysroot' argument in WEBRTCARGS based on whether CMAKE_SYSROOT is defined.

```cmake
if (DEFINED CMAKE_SYSROOT)
  set (WEBRTCARGS use_sysroot=true
${WEBRTCARGS})
else()
  set (WEBRTCARGS use_sysroot=false
${WEBRTCARGS})
endif()
```

--------------------------------

### Set Answer

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Sets the answer for an ongoing communication.

```APIDOC
## POST /api/setAnswer

### Description
Sets the answer for an ongoing communication.

### Method
POST

### Endpoint
/api/setAnswer
```

--------------------------------

### Patching Protobuf Port Definition for Apple

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Applies a patch to the protobuf port definition file for Apple platforms by removing the 'constinit' keyword from a macro definition. This is done to resolve compatibility issues.

```cmake
if (APPLE)
  file(READ ${WEBRTCROOT}/src/third_party/protobuf/src/google/protobuf/port_def.inc filecontent)
  string(REGEX REPLACE "define PROTOBUF_CONSTINIT constinit" "define PROTOBUF_CONSTINIT " filecontent "${filecontent}")
  file(WRITE ${WEBRTCROOT}/src/third_party/protobuf/src/google/protobuf/port_def.inc "${filecontent}")
endif()
```

--------------------------------

### Set clang compilation mode

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Enables clang compilation by setting 'is_clang' to true in WEBRTCARGS.

```cmake
set (WEBRTCARGS is_clang=true
${WEBRTCARGS})
```

--------------------------------

### Static Link libatomic on ARMv6

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Conditionally links libatomic statically on ARMv6 processors for GNU compilers.

```cmake
if (CMAKE_CXX_COMPILER_ID STREQUAL "GNU")
  # static link of libatomic on armv6l
  if (CMAKE_SYSTEM_PROCESSOR MATCHES "armv6.*")
    target_link_libraries (${CMAKE_PROJECT_NAME} -Wl,-Bstatic -latomic -Wl,-Bdynamic)
  endif()

  # static link of stdc++ if available
  include(CheckCXXCompilerFlag)
  CHECK_CXX_COMPILER_FLAG("-static-libgcc" CXX_SUPPORTS_STATIC_LIBGCC)
  if (CXX_SUPPORTS_STATIC_LIBGCC)
    target_link_libraries (${CMAKE_PROJECT_NAME} -static-libgcc)
  endif()
  CHECK_CXX_COMPILER_FLAG("-static-libstdc++" CXX_SUPPORTS_STATIC_LIBSTDCPP)
  if (CXX_SUPPORTS_STATIC_LIBSTDCPP)
    target_link_libraries (${CMAKE_PROJECT_NAME} -static-libstdc++)
  endif()
endif()
```

--------------------------------

### Windows Compiler Flags Override

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Overrides CMAKE_CXX_FLAGS_DEBUG, CMAKE_C_FLAGS_DEBUG, CMAKE_CXX_FLAGS_RELEASE, and CMAKE_C_FLAGS_RELEASE to replace /MD with /MT for static runtime linking on Windows.

```cmake
set(CompilerFlags CMAKE_CXX_FLAGS_DEBUG CMAKE_C_FLAGS_DEBUG CMAKE_CXX_FLAGS_RELEASE CMAKE_C_FLAGS_RELEASE)
foreach(CompilerFlag ${CompilerFlags})
  string(REPLACE "/MD" "/MT" ${CompilerFlag} "${${CompilerFlag}}")
endforeach()
```

--------------------------------

### Removing ToStringIfSet from Media Channel

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/CMakeLists.txt

Removes a specific line related to 'ToStringIfSet' from the media channel header file. This modification aims to clean up or alter logging/debugging output.

```cmake
file(READ ${WEBRTCROOT}/src/media/base/media_channel.h filecontent)
string(REGEX REPLACE "ost << ToStringIfSet[^;]*;" "" filecontent "${filecontent}")
file(WRITE ${WEBRTCROOT}/src/media/base/media_channel.h "${filecontent}")
```

--------------------------------

### Close a Call

Source: https://github.com/mpromonet/webrtc-streamer/blob/master/docs/api.md

Closes an active communication call.

```APIDOC
## POST /api/hangup

### Description
Closes an active communication call.

### Method
POST

### Endpoint
/api/hangup
```

=== COMPLETE CONTENT === This response contains all available snippets from this library. No additional content exists. Do not make further requests.