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[WIP] Add voice call example
This is useful for testing the CallManager and the upcoming DTLS-SRTP support.
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‎examples/CMakeLists.txt

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

‎examples/audio_call/AudioCall.cpp

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

‎examples/audio_call/CMakeLists.txt

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

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