diff --git a/rtc/src/rtcPeerHandler.cpp b/rtc/src/rtcPeerHandler.cpp
new file mode 100644
index 00000000..9bfc6466
--- /dev/null
+++ b/rtc/src/rtcPeerHandler.cpp
@@ -0,0 +1,83 @@
+#include "rtcPeerHandler.hpp"
+
+rtcPeerHandler::rtcPeerHandler() {
+ rtc::InitLogger(rtc::LogLevel::Verbose, NULL);
+}
+
+void rtcPeerHandler::initiateConnection(std::string peerIP, int peerPort) {
+ // Socket connection between client and server
+ SOCKET sock = socket(AF_INET, SOCK_DGRAM, 0);
+ sockaddr_in addr;
+ addr.sin_addr.s_addr = inet_addr(peerIP.c_str());
+ addr.sin_port = htons(peerPort);
+ addr.sin_family = AF_INET;
+
+ rtc::Configuration conf;
+ conf.enableIceTcp = false;
+ conf.disableAutoNegotiation = false;
+
+ auto pc = std::make_shared<rtc::PeerConnection>(conf);
+
+ rtc::Description::Audio media("audio",
+ rtc::Description::Direction::SendRecv);
+ media.addOpusCodec(96);
+ media.setBitrate(64);
+
+ auto track = pc->addTrack(media);
+
+ // auto session = std::make_shared<rtc::MediaHandler>();
+
+ // track->setMediaHandler(session);
+
+ rtc::Reliability rtcRel;
+ rtcRel.unordered = true;
+ rtcRel.type = rtc::Reliability::Type::Timed;
+ rtcRel.rexmit = 500;
+
+ rtc::DataChannelInit rtcConf;
+ rtcConf.reliability = rtcRel;
+ rtcConf.negotiated = false;
+
+ pc->onStateChange([](rtc::PeerConnection::State state) {
+ std::cout << "State: " << state << std::endl;
+ if (state == rtc::PeerConnection::State::Disconnected ||
+ state == rtc::PeerConnection::State::Failed ||
+ state == rtc::PeerConnection::State::Closed) {
+ // remove disconnected client
+ }
+ });
+
+ pc->onGatheringStateChange([](rtc::PeerConnection::GatheringState state) {
+ std::cout << "Gathering State: " << state << std::endl;
+ });
+
+ /*std::tuple<rtc::Track*, rtc::RtcpSrReporter*> addAudio(
+
+ const std::shared_ptr<rtc::PeerConnection> pc,
+ const uint8_t payloadType, const uint32_t ssrc, const std::string cname,
+ const std::string msid, const std::function<void(void)> onOpen) {
+ auto audio = Description::Audio(cname);
+ audio.addOpusCodec(payloadType);
+ audio.addSSRC(ssrc, cname, msid, cname);
+ auto track = pc->addTrack(audio);
+ // create RTP configuration
+ auto rtpConfig = make_shared<RtpPacketizationConfig>(
+ ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate);
+ // create packetizer
+ auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);
+ // create opus handler
+ auto opusHandler = make_shared<OpusPacketizationHandler>(packetizer);
+
+ // add RTCP SR handler
+ auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);
+ opusHandler->addToChain(srReporter);
+
+ // set handler
+ track->setMediaHandler(opusHandler);
+ track->onOpen(onOpen);
+ auto trackData = make_shared<ClientTrackData>(track, srReporter);
+ return trackData;
+ }*/
+
+ pc->createDataChannel("Fosscord voice connection", rtcConf);
+}
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