mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-23 11:17:09 +08:00
479 lines
17 KiB
C++
479 lines
17 KiB
C++
#include "WebRtcTransport.h"
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#include <iostream>
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#include "Rtcp/Rtcp.h"
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#include "Rtsp/RtpReceiver.h"
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#define RTX_SSRC_OFFSET 2
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#define RTP_CNAME "zlmediakit-rtp"
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#define RTX_CNAME "zlmediakit-rtx"
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WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
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_dtls_transport = std::make_shared<RTC::DtlsTransport>(poller, this);
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_ice_server = std::make_shared<RTC::IceServer>(this, makeRandStr(4), makeRandStr(24));
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}
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void WebRtcTransport::onDestory(){
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_dtls_transport = nullptr;
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_ice_server = nullptr;
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) {
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onSendSockData((char *) packet->GetData(), packet->GetSize(), (struct sockaddr_in *) tuple);
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}
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void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
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InfoL;
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}
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void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) {
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InfoL;
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}
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void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) {
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InfoL;
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if (_answer_sdp->media[0].role == DtlsRole::passive) {
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_dtls_transport->Run(RTC::DtlsTransport::Role::SERVER);
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} else {
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_dtls_transport->Run(RTC::DtlsTransport::Role::CLIENT);
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}
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}
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void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) {
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InfoL;
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::OnDtlsTransportConnected(
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const RTC::DtlsTransport *dtlsTransport,
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RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
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uint8_t *srtpLocalKey,
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size_t srtpLocalKeyLen,
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uint8_t *srtpRemoteKey,
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size_t srtpRemoteKeyLen,
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std::string &remoteCert) {
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InfoL;
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_srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);
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_srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
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onStartWebRTC();
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}
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void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
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onSendSockData((char *)data, len);
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){
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auto tuple = _ice_server->GetSelectedTuple();
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assert(tuple);
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onSendSockData(buf, len, (struct sockaddr_in *) tuple, flush);
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}
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const RtcSession& WebRtcTransport::getSdp(SdpType type) const{
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switch (type) {
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case SdpType::offer: return *_offer_sdp;
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case SdpType::answer: return *_answer_sdp;
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default: throw std::invalid_argument("不识别的sdp类型");
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}
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}
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string getFingerprint(const string &algorithm_str, const std::shared_ptr<RTC::DtlsTransport> &transport){
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auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str);
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for (auto &finger_prints : transport->GetLocalFingerprints()) {
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if (finger_prints.algorithm == algorithm) {
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return finger_prints.value;
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}
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}
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throw std::invalid_argument(StrPrinter << "不支持的加密算法:" << algorithm_str);
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}
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void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){
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//设置远端dtls签名
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RTC::DtlsTransport::Fingerprint remote_fingerprint;
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remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm);
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remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash;
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_dtls_transport->SetRemoteFingerprint(remote_fingerprint);
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}
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void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp) const{
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for (auto &m : sdp.media) {
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if (m.type != TrackApplication && !m.rtcp_mux) {
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throw std::invalid_argument("只支持rtcp-mux模式");
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}
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}
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if (sdp.group.mids.empty()) {
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throw std::invalid_argument("只支持group BUNDLE模式");
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}
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}
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std::string WebRtcTransport::getAnswerSdp(const string &offer){
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//// 解析offer sdp ////
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_offer_sdp = std::make_shared<RtcSession>();
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_offer_sdp->loadFrom(offer);
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onCheckSdp(SdpType::offer, *_offer_sdp);
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setRemoteDtlsFingerprint(*_offer_sdp);
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//// sdp 配置 ////
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SdpAttrFingerprint fingerprint;
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fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
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fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
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RtcConfigure configure;
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configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
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onRtcConfigure(configure);
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//// 生成answer sdp ////
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_answer_sdp = configure.createAnswer(*_offer_sdp);
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onCheckSdp(SdpType::answer, *_answer_sdp);
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auto str = _answer_sdp->toString();
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TraceL << "\r\n" << str;
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return str;
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}
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bool is_dtls(char *buf) {
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return ((*buf > 19) && (*buf < 64));
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}
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bool is_rtp(char *buf) {
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RtpHeader *header = (RtpHeader *) buf;
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return ((header->pt < 64) || (header->pt >= 96));
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}
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bool is_rtcp(char *buf) {
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RtpHeader *header = (RtpHeader *) buf;
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return ((header->pt >= 64) && (header->pt < 96));
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}
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void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple) {
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if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
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RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len);
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if (packet == nullptr) {
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WarnL << "parse stun error" << std::endl;
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return;
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}
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_ice_server->ProcessStunPacket(packet, tuple);
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return;
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}
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if (is_dtls(buf)) {
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_dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
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return;
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}
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if (is_rtp(buf)) {
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if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
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onRtp(buf, len);
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} else {
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WarnL;
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}
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return;
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}
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if (is_rtcp(buf)) {
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if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
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onRtcp(buf, len);
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} else {
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WarnL;
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}
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return;
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}
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}
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void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) {
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const uint8_t *p = (uint8_t *) buf;
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bool ret = false;
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if (_srtp_session_send) {
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ret = _srtp_session_send->EncryptRtp(&p, &len, pt);
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}
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if (ret) {
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onSendSockData((char *) p, len, flush);
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}
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}
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void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){
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const uint8_t *p = (uint8_t *) buf;
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bool ret = false;
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if (_srtp_session_send) {
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ret = _srtp_session_send->EncryptRtcp(&p, &len);
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}
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if (ret) {
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onSendSockData((char *) p, len, flush);
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}
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}
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///////////////////////////////////////////////////////////////////////////////////
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WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &poller){
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WebRtcTransportImp::Ptr ret(new WebRtcTransportImp(poller), [](WebRtcTransportImp *ptr){
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ptr->onDestory();
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delete ptr;
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});
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return ret;
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}
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WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
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_socket = Socket::createSocket(poller, false);
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//随机端口，绑定全部网卡
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_socket->bindUdpSock(0);
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_socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
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inputSockData(buf->data(), buf->size(), addr);
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});
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}
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void WebRtcTransportImp::onDestory() {
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WebRtcTransport::onDestory();
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}
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void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) {
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assert(src);
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_src = src;
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}
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void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
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auto ptr = BufferRaw::create();
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ptr->assign(buf, len);
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_socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush);
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}
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///////////////////////////////////////////////////////////////////
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bool WebRtcTransportImp::canSendRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly;
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}
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bool WebRtcTransportImp::canRecvRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly;
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}
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void WebRtcTransportImp::onStartWebRTC() {
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if (canRecvRtp()) {
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_push_src = std::make_shared<RtspMediaSourceImp>(DEFAULT_VHOST, "live", "push");
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auto rtsp_sdp = getSdp(SdpType::answer).toRtspSdp();
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_push_src->setSdp(rtsp_sdp);
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for (auto &m : getSdp(SdpType::offer).media) {
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if (m.type == TrackVideo) {
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_recv_video_ssrc = m.rtp_ssrc.ssrc;
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}
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for (auto &plan : m.plan) {
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auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt);
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if (!hit_pan) {
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continue;
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}
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//获取offer端rtp的ssrc和pt相关信息
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auto &ref = _rtp_receiver[plan.pt];
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_ssrc_info[m.rtp_ssrc.ssrc] = &ref;
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ref.plan = &plan;
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ref.media = &m;
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ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
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ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
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ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
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ref.receiver = std::make_shared<RtpReceiverImp>([&ref, this](RtpPacket::Ptr rtp) {
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onSortedRtp(ref, std::move(rtp));
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}, [ref, this](const RtpPacket::Ptr &rtp) {
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onBeforeSortedRtp(ref, rtp);
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});
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}
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}
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}
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if (canSendRtp()) {
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_reader = _src->getRing()->attach(_socket->getPoller(), true);
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weak_ptr<WebRtcTransportImp> weak_self = shared_from_this();
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_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
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auto strongSelf = weak_self.lock();
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if (!strongSelf) {
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return;
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}
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size_t i = 0;
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pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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strongSelf->onSendRtp(rtp, ++i == pkt->size());
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});
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});
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}
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}
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void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
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WebRtcTransport::onCheckSdp(type, sdp);
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if (type != SdpType::answer || !canSendRtp()) {
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return;
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}
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//添加answer sdp的ssrc信息，并且记录发送rtp的pt
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for (auto &m : sdp.media) {
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if (m.type == TrackApplication) {
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continue;
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}
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m.rtp_ssrc.ssrc = _src->getSsrc(m.type);
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m.rtp_ssrc.cname = RTP_CNAME;
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//todo 先屏蔽rtx，因为chrome报错
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if (false && m.getRelatedRtxPlan(m.plan[0].pt)) {
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m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc;
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m.rtx_ssrc.cname = RTX_CNAME;
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}
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auto rtsp_media = _rtsp_send_sdp.getMedia(m.type);
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if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
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_send_rtp_pt[m.type] = m.plan[0].pt;
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}
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}
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}
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void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
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WebRtcTransport::onRtcConfigure(configure);
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_rtsp_send_sdp.loadFrom(_src->getSdp(), false);
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//根据rtsp流的相关信息，设置rtc最佳编码
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for (auto &m : _rtsp_send_sdp.media) {
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switch (m.type) {
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case TrackVideo: {
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configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec));
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break;
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}
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case TrackAudio: {
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configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec));
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break;
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}
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default:
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break;
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}
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}
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//添加接收端口candidate信息
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configure.addCandidate(*getIceCandidate());
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}
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SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
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auto candidate = std::make_shared<SdpAttrCandidate>();
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candidate->foundation = "udpcandidate";
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//rtp端口
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candidate->component = 1;
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candidate->transport = "udp";
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//优先级，单candidate时随便
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candidate->priority = 100;
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//todo 此处修改为配置文件
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candidate->address = SockUtil::get_local_ip();
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candidate->port = _socket->get_local_port();
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candidate->type = "host";
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return candidate;
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}
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///////////////////////////////////////////////////////////////////
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class RtpReceiverImp : public RtpReceiver {
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public:
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RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb, function<void(const RtpPacket::Ptr &rtp)> cb_before = nullptr){
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_on_sort = std::move(cb);
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_on_before_sort = std::move(cb_before);
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}
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~RtpReceiverImp() override = default;
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bool inputRtp(TrackType type, int samplerate, uint8_t *ptr, size_t len){
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return handleOneRtp((int) type, type, samplerate, ptr, len);
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}
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protected:
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void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override {
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_on_sort(std::move(rtp));
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}
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void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override {
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if (_on_before_sort) {
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_on_before_sort(rtp);
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}
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}
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private:
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function<void(RtpPacket::Ptr rtp)> _on_sort;
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function<void(const RtpPacket::Ptr &rtp)> _on_before_sort;
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};
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void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len);
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for (auto rtcp : rtcps) {
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switch ((RtcpType) rtcp->pt) {
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case RtcpType::RTCP_SR : {
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//对方汇报rtp发送情况
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RtcpSR *sr = (RtcpSR *) rtcp;
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auto it = _ssrc_info.find(sr->ssrc);
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if (it != _ssrc_info.end()) {
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it->second->rtcp_context_recv->onRtcp(sr);
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auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc);
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sendRtcpPacket(rr->data(), rr->size(), true);
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InfoL << "send rtcp rr";
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}
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break;
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}
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case RtcpType::RTCP_RR : {
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//对方汇报rtp接收情况
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RtcpRR *rr = (RtcpRR *) rtcp;
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auto it = _ssrc_info.find(rr->ssrc);
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if (it != _ssrc_info.end()) {
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auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc);
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sendRtcpPacket(sr->data(), sr->size(), true);
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InfoL << "send rtcp sr";
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}
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break;
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}
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case RtcpType::RTCP_BYE : {
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//todo 此处应该销毁对象
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break;
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}
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case RtcpType::RTCP_PSFB: {
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// InfoL << rtcp->dumpString();
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break;
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}
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default: break;
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}
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}
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}
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void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
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RtpHeader *rtp = (RtpHeader *) buf;
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//根据接收到的rtp的pt信息，找到该流的信息
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auto it = _rtp_receiver.find(rtp->pt);
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if (it == _rtp_receiver.end()) {
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WarnL;
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return;
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}
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auto &info = it->second;
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//解析并排序rtp
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info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
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if(!info.is_common_rtp){
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//todo rtx/red/ulpfec类型的rtp先未处理
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WarnL;
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return;
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}
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if (_pli_ticker.elapsedTime() > 2000) {
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//todo 定期发送pli
|
||
_pli_ticker.resetTime();
|
||
auto pli = RtcpPli::create();
|
||
pli->ssrc = htonl(0);
|
||
pli->ssrc_media = htonl(_recv_video_ssrc);
|
||
sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true);
|
||
InfoL << "send pli";
|
||
}
|
||
_push_src->onWrite(std::move(rtp), false);
|
||
}
|
||
|
||
void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
|
||
//统计rtp收到的情况，好做rr汇报
|
||
info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
|
||
}
|
||
|
||
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
|
||
auto &pt = _send_rtp_pt[rtp->type];
|
||
if (!pt) {
|
||
//忽略，对方不支持该编码类型
|
||
return;
|
||
}
|
||
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
|
||
//统计rtp发送情况，好做sr汇报
|
||
_rtp_receiver[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
|
||
}
|