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server/rtc/src/rtcPeerHandler.cpp
Flam3rboy a92c6637ce rtc
2021-08-12 20:33:42 +02:00

84 lines
2.8 KiB
C++

#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);
}