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Ethan
Ethan

Posted on • Originally published at ethan-dev.com

Streaming Camera with C++ WebRTC GStreamer

#cpp #webrtc #tutorial #javascript

Introduction

Hello! 😎

In this advanced WebRTC tutorial I will show you how to stream your camera to a HTML page using WebRTC, GStreamer and C++. We will be using boost to handle the signaling. By the end of this tutorial you should have a simple understanding on WebRTC GStreamer. 👀

Requirements

  • GStreamer and its development libraries
  • Boost libraries
  • CMake for building the project
  • A C++ compiler
  • Basic C++ Knowledge

Creating the Project

First we need a place to house our projects files, create a new directory like so:

 mkdir webrtc-stream && cd webrtc-stream
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First we need to create a build file in order to build the completed project, create a new file called "CMakeLists.txt" and populate it with the following:

 cmake_minimum_required(VERSION 3.10) # Set the project name and version project(webrtc_server VERSION 1.0) # Specify the C++ standard set(CMAKE_CXX_STANDARD 14) set(CMAKE_CXX_STANDARD_REQUIRED True) # Find required packages find_package(PkgConfig REQUIRED) pkg_check_modules(GST REQUIRED gstreamer-1.0 gstreamer-webrtc-1.0 gstreamer-sdp-1.0) find_package(Boost 1.65 REQUIRED COMPONENTS system filesystem json) # Include directories include_directories(${GST_INCLUDE_DIRS} ${Boost_INCLUDE_DIRS}) # Add the executable add_executable(webrtc_server main.cpp) # Link libraries target_link_libraries(webrtc_server ${GST_LIBRARIES} Boost::system Boost::filesystem Boost::json) # Set properties set_target_properties(webrtc_server PROPERTIES CXX_STANDARD 14 CXX_STANDARD_REQUIRED ON ) # Specify additional directories for the linker link_directories(${GST_LIBRARY_DIRS}) # Print project info message(STATUS "Project: ${PROJECT_NAME}") message(STATUS "Version: ${PROJECT_VERSION}") message(STATUS "C++ Standard: ${CMAKE_CXX_STANDARD}") message(STATUS "Boost Libraries: ${Boost_LIBRARIES}") message(STATUS "GStreamer Libraries: ${GST_LIBRARIES}")
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The above links all the required libraries together in order to build the code into an executable that can be executed.

Now we can get on to coding the project. 🥸

Coding the Project

Now we can start coding the source code for the project, create a new file called "main.cpp", we will start by importing the necessary headers for GStreamer, WebRTC, Boost and standard libraries:

 #define GST_USE_UNSTABLE_API #include <gst/gst.h> #include <gst/webrtc/webrtc.h> #include <boost/beast.hpp> #include <boost/asio.hpp> #include <boost/json.hpp> #include <iostream> #include <thread>  namespace beast = boost::beast; namespace http = beast::http; namespace websocket = beast::websocket; namespace net = boost::asio; using tcp = net::ip::tcp; using namespace boost::json;
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Next we will be define constants that will be used later, mainly the STUN server and port that the server will listen on:

 #define STUN_SERVER "stun://stun.l.google.com:19302" #define SERVER_PORT 8000
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Now we will declare global variables for the GStreamer main loop and pipeline elements:

 GMainLoop *loop; GstElement *pipeline, *webrtcbin;
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Next we will create the functions to handle each of the events. First one being a function that sends ICE candidates to the WebSocket client:

 void send_ice_candidate_message(websocket::stream<tcp::socket>& ws, guint mlineindex, gchar *candidate) { std::cout << "Sending ICE candidate: mlineindex=" << mlineindex << ", candidate=" << candidate << std::endl; object ice_json; ice_json["candidate"] = candidate; ice_json["sdpMLineIndex"] = mlineindex; object msg_json; msg_json["type"] = "candidate"; msg_json["ice"] = ice_json; std::string text = serialize(msg_json); ws.write(net::buffer(text)); std::cout << "ICE candidate sent" << std::endl; }
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The next "on_answer_created" function handles the creation of a WebRTC answer and sends it back to the client:

 void on_answer_created(GstPromise *promise, gpointer user_data) { std::cout << "Answer created" << std::endl; websocket::stream<tcp::socket>* ws = static_cast<websocket::stream<tcp::socket>*>(user_data); GstWebRTCSessionDescription *answer = NULL; const GstStructure *reply = gst_promise_get_reply(promise); gst_structure_get(reply, "answer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL); GstPromise *local_promise = gst_promise_new(); g_signal_emit_by_name(webrtcbin, "set-local-description", answer, local_promise); object sdp_json; sdp_json["type"] = "answer"; sdp_json["sdp"] = gst_sdp_message_as_text(answer->sdp); std::string text = serialize(sdp_json); ws->write(net::buffer(text)); std::cout << "Local description set and answer sent: " << text << std::endl; gst_webrtc_session_description_free(answer); }
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The next function is just a placeholder for handling negotiation events, this event is not needed in this example:

 void on_negotiation_needed(GstElement *webrtc, gpointer user_data) { std::cout << "Negotiation needed" << std::endl; }
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The "on_set_remote_description" function sets the remote description and creates an answer:

 void on_set_remote_description(GstPromise *promise, gpointer user_data) { std::cout << "Remote description set, creating answer" << std::endl; websocket::stream<tcp::socket>* ws = static_cast<websocket::stream<tcp::socket>*>(user_data); GstPromise *answer_promise = gst_promise_new_with_change_func(on_answer_created, ws, NULL); g_signal_emit_by_name(webrtcbin, "create-answer", NULL, answer_promise); }
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The "on_ice_candidate" function handles ICE candidate events and sends them to the WebSocket client:

 void on_ice_candidate(GstElement *webrtc, guint mlineindex, gchar *candidate, gpointer user_data) { std::cout << "ICE candidate generated: mlineindex=" << mlineindex << ", candidate=" << candidate << std::endl; websocket::stream<tcp::socket>* ws = static_cast<websocket::stream<tcp::socket>*>(user_data); send_ice_candidate_message(*ws, mlineindex, candidate); }
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The "handle_websocket_session" function manages the WebSocket connection, setting up the GStreamer pipeline and handling both SDP and ICE messages:

 void handle_websocket_session(tcp::socket socket) { try { websocket::stream<tcp::socket> ws{std::move(socket)}; ws.accept(); std::cout << "WebSocket connection accepted" << std::endl; GstStateChangeReturn ret; GError *error = NULL; pipeline = gst_pipeline_new("pipeline"); GstElement *v4l2src = gst_element_factory_make("v4l2src", "source"); GstElement *videoconvert = gst_element_factory_make("videoconvert", "convert"); GstElement *queue = gst_element_factory_make("queue", "queue"); GstElement *vp8enc = gst_element_factory_make("vp8enc", "encoder"); GstElement *rtpvp8pay = gst_element_factory_make("rtpvp8pay", "pay"); webrtcbin = gst_element_factory_make("webrtcbin", "sendrecv"); if (!pipeline || !v4l2src || !videoconvert || !queue || !vp8enc || !rtpvp8pay || !webrtcbin) { g_printerr("Not all elements could be created.\n"); return; } g_object_set(v4l2src, "device", "/dev/video0", NULL); g_object_set(vp8enc, "deadline", 1, NULL); gst_bin_add_many(GST_BIN(pipeline), v4l2src, videoconvert, queue, vp8enc, rtpvp8pay, webrtcbin, NULL); if (!gst_element_link_many(v4l2src, videoconvert, queue, vp8enc, rtpvp8pay, NULL)) { g_printerr("Elements could not be linked.\n"); gst_object_unref(pipeline); return; } GstPad *rtp_src_pad = gst_element_get_static_pad(rtpvp8pay, "src"); GstPad *webrtc_sink_pad = gst_element_get_request_pad(webrtcbin, "sink_%u"); gst_pad_link(rtp_src_pad, webrtc_sink_pad); gst_object_unref(rtp_src_pad); gst_object_unref(webrtc_sink_pad); g_signal_connect(webrtcbin, "on-negotiation-needed", G_CALLBACK(on_negotiation_needed), &ws); g_signal_connect(webrtcbin, "on-ice-candidate", G_CALLBACK(on_ice_candidate), &ws); ret = gst_element_set_state(pipeline, GST_STATE_PLAYING); if (ret == GST_STATE_CHANGE_FAILURE) { g_printerr("Unable to set the pipeline to the playing state.\n"); gst_object_unref(pipeline); return; } std::cout << "GStreamer pipeline set to playing" << std::endl; for (;;) { beast::flat_buffer buffer; ws.read(buffer); auto text = beast::buffers_to_string(buffer.data()); value jv = parse(text); object obj = jv.as_object(); std::string type = obj["type"].as_string().c_str(); if (type == "offer") { std::cout << "Received offer: " << text << std::endl; std::string sdp = obj["sdp"].as_string().c_str(); GstSDPMessage *sdp_message; gst_sdp_message_new_from_text(sdp.c_str(), &sdp_message); GstWebRTCSessionDescription *offer = gst_webrtc_session_description_new(GST_WEBRTC_SDP_TYPE_OFFER, sdp_message); GstPromise *promise = gst_promise_new_with_change_func(on_set_remote_description, &ws, NULL); g_signal_emit_by_name(webrtcbin, "set-remote-description", offer, promise); gst_webrtc_session_description_free(offer); std::cout << "Setting remote description" << std::endl; } else if (type == "candidate") { std::cout << "Received ICE candidate: " << text << std::endl; object ice = obj["ice"].as_object(); std::string candidate = ice["candidate"].as_string().c_str(); guint sdpMLineIndex = ice["sdpMLineIndex"].as_int64(); g_signal_emit_by_name(webrtcbin, "add-ice-candidate", sdpMLineIndex, candidate.c_str()); std::cout << "Added ICE candidate" << std::endl; } } } catch (beast::system_error const& se) { if (se.code() != websocket::error::closed) { std::cerr << "Error: " << se.code().message() << std::endl; } } catch (std::exception const& e) { std::cerr << "Exception: " << e.what() << std::endl; } }
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The next "start_server" function initializes the server, sccepting TCP connections and spawning new threads to handle each connection:

 void start_server() { try { net::io_context ioc{1}; tcp::acceptor acceptor{ioc, tcp::endpoint{tcp::v4(), SERVER_PORT}}; for (;;) { tcp::socket socket{ioc}; acceptor.accept(socket); std::cout << "Accepted new TCP connection" << std::endl; std::thread{handle_websocket_session, std::move(socket)}.detach(); } } catch (std::exception const& e) { std::cerr << "Exception: " << e.what() << std::endl; } }
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Finally we just need to create the final main function to initialize GStreamer, start the server and run the main loop:

 int main(int argc, char *argv[]) { gst_init(&argc, &argv); loop = g_main_loop_new(NULL, FALSE); std::cout << "Starting WebRTC server" << std::endl; std::thread server_thread(start_server); g_main_loop_run(loop); server_thread.join(); gst_element_set_state(pipeline, GST_STATE_NULL); gst_object_unref(pipeline); g_main_loop_unref(loop); std::cout << "WebRTC server stopped" << std::endl; return 0; }
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Done, now we can finally build the project! 😄

Building the Project

To build the above source code into an executable first create a new directory called build:

 mkdir build && cd build
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Build the project:

 cmake .. make
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If all goes well the project should be built successfully and you should have an executable.

Next we need to create a page to view the stream. 😸

Creating the Frontend

Create a new directory called "public" and in it create a new html file called "index.html" and populate it with the following code:

 <!DOCTYPE html> <html> <head> <title>WebRTC Stream</title> </head> <body> <video id="video" autoplay playsinline muted></video> <script> const video = document.getElementById('video'); const signaling = new WebSocket('ws://localhost:8000/ws'); let pc = new RTCPeerConnection({ iceServers: [{urls: 'stun:stun.l.google.com:19302'}] }); signaling.onmessage = async (event) => { const data = JSON.parse(event.data); console.log('Received signaling message:', data); if (data.type === 'answer') { console.log('Setting remote description with answer'); await pc.setRemoteDescription(new RTCSessionDescription(data)); } else if (data.type === 'candidate') { console.log('Adding ICE candidate:', data.ice); await pc.addIceCandidate(new RTCIceCandidate(data.ice)); } }; pc.onicecandidate = (event) => { if (event.candidate) { console.log('Sending ICE candidate:', event.candidate); signaling.send(JSON.stringify({ type: 'candidate', ice: event.candidate })); } }; pc.ontrack = (event) => { console.log('Received track:', event); if (event.track.kind === 'video') { console.log('Attaching video track to video element'); video.srcObject = event.streams[0]; video.play().catch(error => { console.error('Error playing video:', error); }); video.load(); } }; pc.oniceconnectionstatechange = () => { console.log('ICE connection state:', pc.iceConnectionState); }; pc.onicegatheringstatechange = () => { console.log('ICE gathering state:', pc.iceGatheringState); }; pc.onsignalingstatechange = () => { console.log('Signaling state:', pc.signalingState); }; async function start() { pc.addTransceiver('video', {direction: 'recvonly'}); const offer = await pc.createOffer(); console.log('Created offer:', offer); await pc.setLocalDescription(offer); console.log('Set local description with offer'); signaling.send(JSON.stringify({type: 'offer', sdp: pc.localDescription.sdp})); } start(); </script> </body> </html>
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The above can be explained in my other WebRTC tutorials, but it simple communicates with the signalling server and when a remote stream is received plays the video in the video HTML element.

Done now we can actually run the project! 👍

Running the Project

To run the project simple execute the following command:

 ./webrtc_server
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To run the html page we will use a python module:

 python3 -m http.server 9999
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Navigate your browser to http://localhost:9999 and on load you should see your camera showing in the video element like so:

Image of camera stream

Done! 😁

Considerations

In order to improve the above, I would like to implement the following:

  • Handle multiple viewers
  • Handle receiving a stream from HTML
  • Creating an SFU
  • Recording

Conclusion

In this tutorial I have shown you how to stream your camera using native C++, GStreamer and view the stream in a HTML page. I hope this tutorial has taught you something, I certainly had a lot of fun creating it.

As always you can find the source code for the project on my Github:
https://github.com/ethand91/webrtc-gstreamer

Happy Coding! 😎

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