diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
new file mode 100644
index 00000000..ac707243
--- /dev/null
+++ b/src/WebRTCSession.cpp
@@ -0,0 +1,438 @@
+#include "WebRTCSession.h"
+#include "Logging.h"
+
+extern "C" {
+#include "gst/gst.h"
+#include "gst/sdp/sdp.h"
+
+#define GST_USE_UNSTABLE_API
+#include "gst/webrtc/webrtc.h"
+}
+
+namespace {
+bool gisoffer;
+std::string glocalsdp;
+std::vector<mtx::events::msg::CallCandidates::Candidate> gcandidates;
+
+gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data);
+GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type);
+void generateOffer(GstElement *webrtc);
+void setLocalDescription(GstPromise *promise, gpointer webrtc);
+void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED);
+gboolean onICEGatheringCompletion(gpointer timerid);
+void createAnswer(GstPromise *promise, gpointer webrtc);
+void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
+void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
+}
+
+bool
+WebRTCSession::init(std::string *errorMessage)
+{
+ if (initialised_)
+ return true;
+
+ GError *error = nullptr;
+ if (!gst_init_check(nullptr, nullptr, &error)) {
+ std::string strError = std::string("Failed to initialise GStreamer: ");
+ if (error) {
+ strError += error->message;
+ g_error_free(error);
+ }
+ nhlog::ui()->error(strError);
+ if (errorMessage)
+ *errorMessage = strError;
+ return false;
+ }
+
+ gchar *version = gst_version_string();
+ std::string gstVersion(version);
+ g_free(version);
+ nhlog::ui()->info("Initialised " + gstVersion);
+
+ // GStreamer Plugins:
+ // Base: audioconvert, audioresample, opus, playback, videoconvert, volume
+ // Good: autodetect, rtpmanager, vpx
+ // Bad: dtls, srtp, webrtc
+ // libnice [GLib]: nice
+ initialised_ = true;
+ std::string strError = gstVersion + ": Missing plugins: ";
+ const gchar *needed[] = {"audioconvert", "audioresample", "autodetect", "dtls", "nice",
+ "opus", "playback", "rtpmanager", "srtp", "videoconvert", "vpx", "volume", "webrtc", nullptr};
+ GstRegistry *registry = gst_registry_get();
+ for (guint i = 0; i < g_strv_length((gchar**)needed); i++) {
+ GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
+ if (!plugin) {
+ strError += needed[i];
+ initialised_ = false;
+ continue;
+ }
+ gst_object_unref(plugin);
+ }
+
+ if (!initialised_) {
+ nhlog::ui()->error(strError);
+ if (errorMessage)
+ *errorMessage = strError;
+ }
+ return initialised_;
+}
+
+bool
+WebRTCSession::createOffer()
+{
+ gisoffer = true;
+ glocalsdp.clear();
+ gcandidates.clear();
+ return startPipeline(111); // a dynamic opus payload type
+}
+
+bool
+WebRTCSession::acceptOffer(const std::string& sdp)
+{
+ nhlog::ui()->debug("Received offer:\n{}", sdp);
+ gisoffer = false;
+ glocalsdp.clear();
+ gcandidates.clear();
+
+ // eg a=rtpmap:111 opus/48000/2
+ int opusPayloadType = 0;
+ if (auto e = sdp.find("opus"); e == std::string::npos) {
+ nhlog::ui()->error("WebRTC: remote offer - opus media attribute missing");
+ return false;
+ }
+ else {
+ if (auto s = sdp.rfind(':', e); s == std::string::npos) {
+ nhlog::ui()->error("WebRTC: remote offer - unable to determine opus payload type");
+ return false;
+ }
+ else {
+ ++s;
+ try {
+ opusPayloadType = std::stoi(std::string(sdp, s, e - s));
+ }
+ catch(...) {
+ nhlog::ui()->error("WebRTC: remote offer - unable to determine opus payload type");
+ return false;
+ }
+ }
+ }
+
+ GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
+ if (!offer)
+ return false;
+
+ if (!startPipeline(opusPayloadType))
+ return false;
+
+ // set-remote-description first, then create-answer
+ GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
+ g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
+ gst_webrtc_session_description_free(offer);
+ return true;
+}
+
+bool
+WebRTCSession::startPipeline(int opusPayloadType)
+{
+ if (isActive())
+ return false;
+
+ if (!createPipeline(opusPayloadType))
+ return false;
+
+ webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
+
+ if (!stunServer_.empty()) {
+ nhlog::ui()->info("WebRTC: Setting stun server: {}", stunServer_);
+ g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
+ }
+ addTurnServers();
+
+ // generate the offer when the pipeline goes to PLAYING
+ if (gisoffer)
+ g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr);
+
+ // on-ice-candidate is emitted when a local ICE candidate has been gathered
+ g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
+
+ // incoming streams trigger pad-added
+ gst_element_set_state(pipe_, GST_STATE_READY);
+ g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
+
+ // webrtcbin lifetime is the same as that of the pipeline
+ gst_object_unref(webrtc_);
+
+ // start the pipeline
+ GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
+ if (ret == GST_STATE_CHANGE_FAILURE) {
+ nhlog::ui()->error("WebRTC: unable to start pipeline");
+ gst_object_unref(pipe_);
+ pipe_ = nullptr;
+ webrtc_ = nullptr;
+ return false;
+ }
+
+ GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
+ gst_bus_add_watch(bus, newBusMessage, this);
+ gst_object_unref(bus);
+ emit pipelineChanged(true);
+ return true;
+}
+
+#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
+
+bool
+WebRTCSession::createPipeline(int opusPayloadType)
+{
+ std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
+ "autoaudiosrc ! volume name=srclevel ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
+ "queue ! " RTP_CAPS_OPUS + std::to_string(opusPayloadType) + " ! webrtcbin.");
+
+ webrtc_ = nullptr;
+ GError *error = nullptr;
+ pipe_ = gst_parse_launch(pipeline.c_str(), &error);
+ if (error) {
+ nhlog::ui()->error("WebRTC: Failed to parse pipeline: {}", error->message);
+ g_error_free(error);
+ if (pipe_) {
+ gst_object_unref(pipe_);
+ pipe_ = nullptr;
+ }
+ return false;
+ }
+ return true;
+}
+
+bool
+WebRTCSession::acceptAnswer(const std::string &sdp)
+{
+ nhlog::ui()->debug("WebRTC: Received sdp:\n{}", sdp);
+ if (!isActive())
+ return false;
+
+ GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
+ if (!answer)
+ return false;
+
+ g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
+ gst_webrtc_session_description_free(answer);
+ return true;
+}
+
+void
+WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate>& candidates)
+{
+ if (isActive()) {
+ for (const auto& c : candidates)
+ g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
+ }
+}
+
+bool
+WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
+{
+ if (!isActive())
+ return false;
+
+ GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
+ if (!srclevel)
+ return false;
+
+ gboolean muted;
+ g_object_get(srclevel, "mute", &muted, nullptr);
+ g_object_set(srclevel, "mute", !muted, nullptr);
+ gst_object_unref(srclevel);
+ isMuted = !muted;
+ return true;
+}
+
+void
+WebRTCSession::end()
+{
+ if (pipe_) {
+ gst_element_set_state(pipe_, GST_STATE_NULL);
+ gst_object_unref(pipe_);
+ pipe_ = nullptr;
+ }
+ webrtc_ = nullptr;
+ emit pipelineChanged(false);
+}
+
+void
+WebRTCSession::addTurnServers()
+{
+ if (!webrtc_)
+ return;
+
+ for (const auto &uri : turnServers_) {
+ gboolean res;
+ g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&res));
+ if (res)
+ nhlog::ui()->info("WebRTC: Set TURN server: {}", uri);
+ else
+ nhlog::ui()->error("WebRTC: Failed to set TURN server: {}", uri);
+ }
+}
+
+namespace {
+
+gboolean
+newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
+{
+ WebRTCSession *session = (WebRTCSession*)user_data;
+ switch (GST_MESSAGE_TYPE(msg)) {
+ case GST_MESSAGE_EOS:
+ session->end();
+ break;
+ case GST_MESSAGE_ERROR:
+ GError *error;
+ gchar *debug;
+ gst_message_parse_error(msg, &error, &debug);
+ nhlog::ui()->error("WebRTC: Error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
+ g_clear_error(&error);
+ g_free(debug);
+ session->end();
+ break;
+ default:
+ break;
+ }
+ return TRUE;
+}
+
+GstWebRTCSessionDescription*
+parseSDP(const std::string &sdp, GstWebRTCSDPType type)
+{
+ GstSDPMessage *msg;
+ gst_sdp_message_new(&msg);
+ if (gst_sdp_message_parse_buffer((guint8*)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
+ return gst_webrtc_session_description_new(type, msg);
+ }
+ else {
+ nhlog::ui()->error("WebRTC: Failed to parse remote session description");
+ gst_object_unref(msg);
+ return nullptr;
+ }
+}
+
+void
+generateOffer(GstElement *webrtc)
+{
+ // create-offer first, then set-local-description
+ GstPromise *promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
+ g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
+}
+
+void
+setLocalDescription(GstPromise *promise, gpointer webrtc)
+{
+ const GstStructure *reply = gst_promise_get_reply(promise);
+ gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
+ GstWebRTCSessionDescription *gstsdp = nullptr;
+ gst_structure_get(reply, isAnswer ? "answer" : "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &gstsdp, nullptr);
+ gst_promise_unref(promise);
+ g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
+
+ gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
+ glocalsdp = std::string(sdp);
+ g_free(sdp);
+ gst_webrtc_session_description_free(gstsdp);
+
+ nhlog::ui()->debug("WebRTC: Local description set ({}):\n{}", isAnswer ? "answer" : "offer", glocalsdp);
+}
+
+void
+addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED)
+{
+ gcandidates.push_back({"audio", (uint16_t)mlineIndex, candidate});
+
+ // GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
+ // fixed in v1.18
+ // use a 100ms timeout in the meantime
+ static guint timerid = 0;
+ if (timerid)
+ g_source_remove(timerid);
+
+ timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
+}
+
+gboolean
+onICEGatheringCompletion(gpointer timerid)
+{
+ *(guint*)(timerid) = 0;
+ if (gisoffer)
+ emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
+ else
+ emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
+
+ return FALSE;
+}
+
+void
+createAnswer(GstPromise *promise, gpointer webrtc)
+{
+ // create-answer first, then set-local-description
+ gst_promise_unref(promise);
+ promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
+ g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
+}
+
+void
+addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
+{
+ if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
+ return;
+
+ GstElement *decodebin = gst_element_factory_make("decodebin", nullptr);
+ g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
+ gst_bin_add(GST_BIN(pipe), decodebin);
+ gst_element_sync_state_with_parent(decodebin);
+ GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
+ if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad)))
+ nhlog::ui()->error("WebRTC: Unable to link new pad");
+ gst_object_unref(sinkpad);
+}
+
+void
+linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
+{
+ GstCaps *caps = gst_pad_get_current_caps(newpad);
+ if (!caps)
+ return;
+
+ const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
+ gst_caps_unref(caps);
+
+ GstPad *queuepad = nullptr;
+ GstElement *queue = gst_element_factory_make("queue", nullptr);
+
+ if (g_str_has_prefix(name, "audio")) {
+ GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement *resample = gst_element_factory_make("audioresample", nullptr);
+ GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
+ gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
+ gst_element_sync_state_with_parent(queue);
+ gst_element_sync_state_with_parent(convert);
+ gst_element_sync_state_with_parent(resample);
+ gst_element_sync_state_with_parent(sink);
+ gst_element_link_many(queue, convert, resample, sink, nullptr);
+ queuepad = gst_element_get_static_pad(queue, "sink");
+ }
+ else if (g_str_has_prefix(name, "video")) {
+ GstElement *convert = gst_element_factory_make("videoconvert", nullptr);
+ GstElement *sink = gst_element_factory_make("autovideosink", nullptr);
+ gst_bin_add_many(GST_BIN(pipe), queue, convert, sink, nullptr);
+ gst_element_sync_state_with_parent(queue);
+ gst_element_sync_state_with_parent(convert);
+ gst_element_sync_state_with_parent(sink);
+ gst_element_link_many(queue, convert, sink, nullptr);
+ queuepad = gst_element_get_static_pad(queue, "sink");
+ }
+
+ if (queuepad) {
+ if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
+ nhlog::ui()->error("WebRTC: Unable to link new pad");
+ gst_object_unref(queuepad);
+ }
+}
+
+}
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