From a92c6637cea0ce595c712c3d445b57a1184e833e Mon Sep 17 00:00:00 2001
From: Flam3rboy <34555296+Flam3rboy@users.noreply.github.com>
Date: Thu, 12 Aug 2021 20:33:42 +0200
Subject: :sparkles: rtc

---
 rtc/src/rtcPeerHandler.cpp | 83 ++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 83 insertions(+)
 create mode 100644 rtc/src/rtcPeerHandler.cpp

(limited to 'rtc/src/rtcPeerHandler.cpp')

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);
+}
-- 
cgit 1.5.1