mirror of
https://github.com/Nheko-Reborn/nheko.git
synced 2024-11-22 11:00:48 +03:00
Handle ICE failure
This commit is contained in:
parent
57d5a3d31f
commit
43ec0c0624
7 changed files with 131 additions and 73 deletions
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@ -123,25 +123,32 @@ ActiveCallBar::update(WebRTCSession::State state)
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{
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switch (state) {
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case WebRTCSession::State::INITIATING:
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show();
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stateLabel_->setText("Initiating call...");
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break;
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case WebRTCSession::State::INITIATED:
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show();
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stateLabel_->setText("Call initiated...");
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break;
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case WebRTCSession::State::OFFERSENT:
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show();
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stateLabel_->setText("Calling...");
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break;
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case WebRTCSession::State::CONNECTING:
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show();
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stateLabel_->setText("Connecting...");
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break;
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case WebRTCSession::State::CONNECTED:
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show();
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callStartTime_ = QDateTime::currentSecsSinceEpoch();
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timer_->start(1000);
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stateLabel_->setText("Voice call:");
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durationLabel_->setText("00:00");
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durationLabel_->show();
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break;
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case WebRTCSession::State::ICEFAILED:
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case WebRTCSession::State::DISCONNECTED:
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hide();
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timer_->stop();
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callPartyLabel_->setText(QString());
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stateLabel_->setText(QString());
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@ -11,9 +11,10 @@
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#include "MatrixClient.h"
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#include "UserSettingsPage.h"
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#include "WebRTCSession.h"
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#include "dialogs/AcceptCall.h"
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#include "mtx/responses/turn_server.hpp"
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Q_DECLARE_METATYPE(std::vector<mtx::events::msg::CallCandidates::Candidate>)
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Q_DECLARE_METATYPE(mtx::events::msg::CallCandidates::Candidate)
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Q_DECLARE_METATYPE(mtx::responses::TurnServer)
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@ -24,6 +25,11 @@ using namespace mtx::events::msg;
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// https://github.com/vector-im/riot-web/issues/10173
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#define STUN_SERVER "stun://turn.matrix.org:3478"
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namespace {
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std::vector<std::string>
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getTurnURIs(const mtx::responses::TurnServer &turnServer);
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}
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CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
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: QObject(),
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session_(WebRTCSession::instance()),
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@ -80,15 +86,23 @@ CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
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// Request new credentials close to expiry
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// See https://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
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turnServer_ = res;
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turnURIs_ = getTurnURIs(res);
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turnServerTimer_.setInterval(res.ttl * 1000 * 0.9);
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});
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connect(&session_, &WebRTCSession::stateChanged, this,
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[this](WebRTCSession::State state) {
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if (state == WebRTCSession::State::DISCONNECTED)
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if (state == WebRTCSession::State::DISCONNECTED) {
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playRingtone("qrc:/media/media/callend.ogg", false);
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});
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}
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else if (state == WebRTCSession::State::ICEFAILED) {
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QString error("Call connection failed.");
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if (turnURIs_.empty())
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error += " Your homeserver has no configured TURN server.";
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emit ChatPage::instance()->showNotification(error);
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hangUp(CallHangUp::Reason::ICEFailed);
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}
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});
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connect(&player_, &QMediaPlayer::mediaStatusChanged, this,
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[this](QMediaPlayer::MediaStatus status) {
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@ -116,8 +130,8 @@ CallManager::sendInvite(const QString &roomid)
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}
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roomid_ = roomid;
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setTurnServers();
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session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
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session_.setTurnServers(turnURIs_);
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generateCallID();
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nhlog::ui()->debug("WebRTC: call id: {} - creating invite", callid_);
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@ -132,11 +146,26 @@ CallManager::sendInvite(const QString &roomid)
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}
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}
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namespace {
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std::string callHangUpReasonString(CallHangUp::Reason reason)
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{
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switch (reason) {
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case CallHangUp::Reason::ICEFailed:
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return "ICE failed";
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case CallHangUp::Reason::InviteTimeOut:
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return "Invite time out";
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default:
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return "User";
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}
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}
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}
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void
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CallManager::hangUp(CallHangUp::Reason reason)
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{
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if (!callid_.empty()) {
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nhlog::ui()->debug("WebRTC: call id: {} - hanging up", callid_);
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nhlog::ui()->debug("WebRTC: call id: {} - hanging up ({})", callid_,
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callHangUpReasonString(reason));
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emit newMessage(roomid_, CallHangUp{callid_, 0, reason});
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endCall();
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}
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@ -221,8 +250,8 @@ CallManager::answerInvite(const CallInvite &invite)
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return;
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}
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setTurnServers();
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session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
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session_.setTurnServers(turnURIs_);
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if (!session_.acceptOffer(invite.sdp)) {
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emit ChatPage::instance()->showNotification("Problem setting up call.");
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@ -279,8 +308,9 @@ CallManager::handleEvent(const RoomEvent<CallAnswer> &callAnswerEvent)
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void
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CallManager::handleEvent(const RoomEvent<CallHangUp> &callHangUpEvent)
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{
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nhlog::ui()->debug("WebRTC: call id: {} - incoming CallHangUp from {}",
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callHangUpEvent.content.call_id, callHangUpEvent.sender);
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nhlog::ui()->debug("WebRTC: call id: {} - incoming CallHangUp ({}) from {}",
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callHangUpEvent.content.call_id, callHangUpReasonString(callHangUpEvent.content.reason),
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callHangUpEvent.sender);
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if (callid_ == callHangUpEvent.content.call_id) {
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MainWindow::instance()->hideOverlay();
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@ -319,35 +349,6 @@ CallManager::retrieveTurnServer()
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});
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}
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void
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CallManager::setTurnServers()
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{
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// gstreamer expects: turn(s)://username:password@host:port?transport=udp(tcp)
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// where username and password are percent-encoded
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std::vector<std::string> uris;
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for (const auto &uri : turnServer_.uris) {
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if (auto c = uri.find(':'); c == std::string::npos) {
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nhlog::ui()->error("Invalid TURN server uri: {}", uri);
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continue;
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}
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else {
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std::string scheme = std::string(uri, 0, c);
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if (scheme != "turn" && scheme != "turns") {
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nhlog::ui()->error("Invalid TURN server uri: {}", uri);
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continue;
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}
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QString encodedUri = QString::fromStdString(scheme) + "://" +
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QUrl::toPercentEncoding(QString::fromStdString(turnServer_.username)) + ":" +
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QUrl::toPercentEncoding(QString::fromStdString(turnServer_.password)) + "@" +
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QString::fromStdString(std::string(uri, ++c));
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uris.push_back(encodedUri.toStdString());
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}
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}
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if (!uris.empty())
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session_.setTurnServers(uris);
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}
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void
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CallManager::playRingtone(const QString &ringtone, bool repeat)
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{
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@ -364,3 +365,34 @@ CallManager::stopRingtone()
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{
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player_.setPlaylist(nullptr);
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}
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namespace {
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std::vector<std::string>
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getTurnURIs(const mtx::responses::TurnServer &turnServer)
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{
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// gstreamer expects: turn(s)://username:password@host:port?transport=udp(tcp)
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// where username and password are percent-encoded
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std::vector<std::string> ret;
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for (const auto &uri : turnServer.uris) {
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if (auto c = uri.find(':'); c == std::string::npos) {
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nhlog::ui()->error("Invalid TURN server uri: {}", uri);
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continue;
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}
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else {
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std::string scheme = std::string(uri, 0, c);
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if (scheme != "turn" && scheme != "turns") {
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nhlog::ui()->error("Invalid TURN server uri: {}", uri);
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continue;
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}
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QString encodedUri = QString::fromStdString(scheme) + "://" +
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QUrl::toPercentEncoding(QString::fromStdString(turnServer.username)) + ":" +
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QUrl::toPercentEncoding(QString::fromStdString(turnServer.password)) + "@" +
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QString::fromStdString(std::string(uri, ++c));
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ret.push_back(encodedUri.toStdString());
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}
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}
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return ret;
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}
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}
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@ -11,7 +11,10 @@
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#include "mtx/events/collections.hpp"
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#include "mtx/events/voip.hpp"
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#include "mtx/responses/turn_server.hpp"
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namespace mtx::responses {
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struct TurnServer;
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}
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class UserSettings;
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class WebRTCSession;
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@ -51,7 +54,7 @@ private:
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std::string callid_;
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const uint32_t timeoutms_ = 120000;
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std::vector<mtx::events::msg::CallCandidates::Candidate> remoteICECandidates_;
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mtx::responses::TurnServer turnServer_;
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std::vector<std::string> turnURIs_;
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QTimer turnServerTimer_;
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QSharedPointer<UserSettings> settings_;
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QMediaPlayer player_;
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@ -65,7 +68,6 @@ private:
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void answerInvite(const mtx::events::msg::CallInvite&);
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void generateCallID();
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void endCall();
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void setTurnServers();
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void playRingtone(const QString &ringtone, bool repeat);
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void stopRingtone();
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};
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@ -137,15 +137,6 @@ ChatPage::ChatPage(QSharedPointer<UserSettings> userSettings, QWidget *parent)
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activeCallBar_->hide();
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connect(
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&callManager_, &CallManager::newCallParty, activeCallBar_, &ActiveCallBar::setCallParty);
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connect(&WebRTCSession::instance(),
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&WebRTCSession::stateChanged,
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this,
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[this](WebRTCSession::State state) {
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if (state == WebRTCSession::State::DISCONNECTED)
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activeCallBar_->hide();
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else
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activeCallBar_->show();
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});
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// Splitter
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splitter->addWidget(sideBar_);
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@ -666,7 +666,8 @@ void
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TextInputWidget::changeCallButtonState(WebRTCSession::State state)
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{
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QIcon icon;
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if (state == WebRTCSession::State::DISCONNECTED) {
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if (state == WebRTCSession::State::ICEFAILED ||
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state == WebRTCSession::State::DISCONNECTED) {
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callBtn_->setToolTip(tr("Place a call"));
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icon.addFile(":/icons/icons/ui/place-call.png");
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} else {
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@ -14,9 +14,9 @@ extern "C" {
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Q_DECLARE_METATYPE(WebRTCSession::State)
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namespace {
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bool gisoffer;
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std::string glocalsdp;
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std::vector<mtx::events::msg::CallCandidates::Candidate> gcandidates;
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bool isoffering_;
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std::string localsdp_;
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std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
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gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data);
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GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type);
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@ -24,6 +24,7 @@ void generateOffer(GstElement *webrtc);
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void setLocalDescription(GstPromise *promise, gpointer webrtc);
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void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED);
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gboolean onICEGatheringCompletion(gpointer timerid);
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void iceConnectionStateChanged(GstElement *webrtcbin, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED);
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void createAnswer(GstPromise *promise, gpointer webrtc);
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void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
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void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
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@ -92,9 +93,9 @@ WebRTCSession::init(std::string *errorMessage)
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bool
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WebRTCSession::createOffer()
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{
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gisoffer = true;
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glocalsdp.clear();
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gcandidates.clear();
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isoffering_ = true;
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localsdp_.clear();
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localcandidates_.clear();
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return startPipeline(111); // a dynamic opus payload type
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}
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@ -105,9 +106,9 @@ WebRTCSession::acceptOffer(const std::string &sdp)
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if (state_ != State::DISCONNECTED)
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return false;
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gisoffer = false;
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glocalsdp.clear();
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gcandidates.clear();
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isoffering_ = false;
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localsdp_.clear();
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localcandidates_.clear();
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int opusPayloadType = getPayloadType(sdp, "opus");
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if (opusPayloadType == -1)
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@ -152,14 +153,20 @@ WebRTCSession::startPipeline(int opusPayloadType)
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gboolean udata;
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g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
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}
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if (turnServers_.empty())
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nhlog::ui()->warn("WebRTC: no TURN server provided");
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// generate the offer when the pipeline goes to PLAYING
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if (gisoffer)
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if (isoffering_)
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g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr);
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// on-ice-candidate is emitted when a local ICE candidate has been gathered
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g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
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// capture ICE failure
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g_signal_connect(webrtc_, "notify::ice-connection-state",
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G_CALLBACK(iceConnectionStateChanged), nullptr);
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// incoming streams trigger pad-added
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gst_element_set_state(pipe_, GST_STATE_READY);
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g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
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@ -229,8 +236,6 @@ WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandi
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nhlog::ui()->debug("WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
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g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
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}
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if (state_ == State::OFFERSENT)
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emit stateChanged(State::CONNECTING);
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}
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}
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@ -357,11 +362,11 @@ setLocalDescription(GstPromise *promise, gpointer webrtc)
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g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
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gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
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glocalsdp = std::string(sdp);
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localsdp_ = std::string(sdp);
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g_free(sdp);
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gst_webrtc_session_description_free(gstsdp);
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nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", glocalsdp);
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nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
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}
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void
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@ -369,12 +374,12 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *
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{
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nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
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if (WebRTCSession::instance().state() == WebRTCSession::State::CONNECTED) {
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if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
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emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
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return;
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}
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gcandidates.push_back({"audio", (uint16_t)mlineIndex, candidate});
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localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
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// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
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// fixed in v1.18
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@ -390,18 +395,36 @@ gboolean
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onICEGatheringCompletion(gpointer timerid)
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{
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*(guint*)(timerid) = 0;
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if (gisoffer) {
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emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
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if (isoffering_) {
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emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
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}
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else {
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emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
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emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
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}
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return FALSE;
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}
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void
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iceConnectionStateChanged(GstElement *webrtc, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED)
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{
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GstWebRTCICEConnectionState newState;
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g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
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switch (newState) {
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case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
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nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
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break;
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case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
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nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
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break;
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default:
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break;
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}
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}
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void
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createAnswer(GstPromise *promise, gpointer webrtc)
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{
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@ -15,10 +15,12 @@ class WebRTCSession : public QObject
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|||
|
||||
public:
|
||||
enum class State {
|
||||
ICEFAILED,
|
||||
DISCONNECTED,
|
||||
INITIATING,
|
||||
INITIATED,
|
||||
OFFERSENT,
|
||||
ANSWERSENT,
|
||||
CONNECTING,
|
||||
CONNECTED
|
||||
};
|
||||
|
@ -30,13 +32,13 @@ public:
|
|||
}
|
||||
|
||||
bool init(std::string *errorMessage = nullptr);
|
||||
State state() const {return state_;}
|
||||
|
||||
bool createOffer();
|
||||
bool acceptOffer(const std::string &sdp);
|
||||
bool acceptAnswer(const std::string &sdp);
|
||||
void acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate>&);
|
||||
|
||||
State state() const {return state_;}
|
||||
bool toggleMuteAudioSrc(bool &isMuted);
|
||||
void end();
|
||||
|
||||
|
|
Loading…
Reference in a new issue