Voice call example + DTLS-SRTP

examples/CMakeLists.txt

@@ -22,4 +22,6 @@ add_simple_example(8_server)
 add_subdirectory(example_5_rpcInterface)
 add_subdirectory(example_6_rpcClient)
 add_subdirectory(example_9_vCard)
-
+if(WITH_GSTREAMER)
+    add_subdirectory(audio_call)
+endif()

examples/audio_call/AudioCall.cpp

@@ -0,0 +1,155 @@
+// SPDX-FileCopyrightText: 2025 Linus Jahn <[email protected]>
+//
+// SPDX-License-Identifier: LGPL-2.1-or-later
+
+#include <QXmppCallManager.h>
+#include <QXmppClient.h>
+#include <QXmppRosterManager.h>
+
+#include <chrono>
+#include <gst/gst.h>
+
+#include <QCoreApplication>
+#include <QTimer>
+#ifdef Q_OS_UNIX
+#include <csignal>
+#endif
+
+using namespace std::chrono_literals;
+
+#ifdef Q_OS_UNIX
+void handleSignal(int signal)
+{
+    if (signal == SIGINT) {
+        // print newline
+        qDebug() << "";
+        if (QCoreApplication::instance()) {
+            QCoreApplication::instance()->quit();
+        }
+    }
+}
+#endif
+
+void setupCallStream(QXmppCall *call)
+{
+    auto *gstPipeline = call->pipeline();
+    auto *stream = call->audioStream();
+
+    qDebug() << "[Call] Setup call stream" << stream->media();
+    if (stream->media() == u"audio") {
+        // output receiving audio
+        stream->setReceivePadCallback([gstPipeline](GstPad *receivePad) {
+            GstElement *output = gst_parse_bin_from_description("audioresample ! audioconvert ! autoaudiosink", true, nullptr);
+            if (!gst_bin_add(GST_BIN(gstPipeline), output)) {
+                qFatal("Failed to add input to pipeline");
+                return;
+            }
+
+            gst_pad_link(receivePad, gst_element_get_static_pad(output, "sink"));
+            gst_element_sync_state_with_parent(output);
+
+            qDebug() << "[Call] receive pad set";
+        });
+
+        // record and send microphone
+        stream->setSendPadCallback([gstPipeline](GstPad *sendPad) {
+            GstElement *output = gst_parse_bin_from_description("autoaudiosrc ! audioconvert ! audioresample ! queue max-size-time=1000000", true, nullptr);
+            if (!gst_bin_add(GST_BIN(gstPipeline), output)) {
+                qFatal("Failed to add input to pipeline");
+                return;
+            }
+
+            gst_pad_link(gst_element_get_static_pad(output, "src"), sendPad);
+            gst_element_sync_state_with_parent(output);
+
+            qDebug() << "[Call] send pad set";
+        });
+    }
+};
+
+int main(int argc, char *argv[])
+{
+    QCoreApplication app(argc, argv);
+
+#ifdef Q_OS_UNIX
+    // set signal handler for SIGINT (CTRL+C)
+    std::signal(SIGINT, handleSignal);
+#endif
+
+    QXmppClient client;
+    auto *rosterManager = client.findExtension<QXmppRosterManager>();
+    auto *callManager = client.addNewExtension<QXmppCallManager>();
+    client.logger()->setLoggingType(QXmppLogger::StdoutLogging);
+    client.logger()->setMessageTypes(QXmppLogger::MessageType::AnyMessage);
+
+    // client config
+    QXmppConfiguration config;
+    config.setJid(qEnvironmentVariable("QXMPP_JID"));
+    config.setPassword(qEnvironmentVariable("QXMPP_PASSWORD"));
+    config.setResourcePrefix("Call");
+
+    // call manager config
+    callManager->setStunServer(QHostAddress(QStringLiteral("stun.nextcloud.com")), 443);
+    // callManager->setTurnServer();
+    // callManager->setTurnUser(client.configuration().jid());
+    // callManager->setTurnUser(client.configuration().password());
+
+    client.connectToServer(config);
+
+    auto setupCall = [&app, callManager](QXmppCall *call) {
+        if (call->audioStream()) {
+            setupCallStream(call);
+        }
+
+        QObject::connect(call, &QXmppCall::streamCreated, call, [call](QXmppCallStream *stream) {
+            setupCallStream(call);
+        });
+
+        QObject::connect(call, &QXmppCall::connected, &app, [=]() {
+            qDebug() << "[Call] Call to" << call->jid() << "connected!";
+        });
+        QObject::connect(call, &QXmppCall::ringing, [=]() {
+            qDebug() << "[Call] Ringing" << call->jid() << "...";
+        });
+        QObject::connect(call, &QXmppCall::finished, [=]() {
+            qDebug() << "[Call] Call with" << call->jid() << "ended. (Deleting)";
+            call->deleteLater();
+        });
+    };
+
+    // on connect
+    QObject::connect(&client, &QXmppClient::connected, &app, [=, &config] {
+        // wait 1 second for presence of other clients to arrive
+        QTimer::singleShot(1s, [=, &config] {
+            // other resources of our account
+            auto otherResources = rosterManager->getResources(config.jidBare());
+            otherResources.removeOne(config.resource());
+            if (otherResources.isEmpty()) {
+                qDebug() << "[Call] No other clients to call on this account.";
+                return;
+            }
+
+            // call first JID
+            auto *call = callManager->call(config.jidBare() + u'/' + otherResources.first());
+            Q_ASSERT(call != nullptr);
+
+            setupCall(call);
+        });
+    });
+
+    // on incoming call
+    QObject::connect(callManager, &QXmppCallManager::callReceived, &app, [=](QXmppCall *call) {
+        qDebug() << "[Call] Received incoming call from" << call->jid() << "-" << "Accepting.";
+        call->accept();
+
+        setupCall(call);
+    });
+
+    // disconnect from server to avoid having multiple open dead sessions when testing
+    QObject::connect(&app, &QCoreApplication::aboutToQuit, &app, [&client]() {
+        qDebug() << "Closing connection...";
+        client.disconnectFromServer();
+    });
+
+    return app.exec();
+}

examples/audio_call/CMakeLists.txt

@@ -0,0 +1,25 @@
+# SPDX-FileCopyrightText: 2025 Linus Jahn <[email protected]>
+#
+# SPDX-License-Identifier: CC0-1.0
+
+find_package(GStreamer REQUIRED)
+find_package(GLIB2 REQUIRED)
+find_package(GObject REQUIRED)
+
+add_executable(audio_call AudioCall.cpp)
+
+target_link_libraries(audio_call
+    PUBLIC
+    ${QXMPP_TARGET}
+    ${GLIB2_LIBRARIES}
+    ${GOBJECT_LIBRARIES}
+    ${GSTREAMER_LIBRARY}
+)
+
+target_include_directories(audio_call
+    PUBLIC
+    ${QXMPP_TARGET}
+    ${GLIB2_INCLUDE_DIR}
+    ${GOBJECT_INCLUDE_DIR}
+    ${GSTREAMER_INCLUDE_DIRS}
+)