ZLMediaKit/webrtc/WebRtcTransport.cpp

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/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include "WebRtcTransport.h"
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#include "Rtcp/Rtcp.h"
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#include "Rtcp/RtcpFCI.h"
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#include "RtpExt.h"
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#include "Rtsp/RtpReceiver.h"
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#include <srtp2/srtp.h>
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#include <iostream>
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#define RTP_SSRC_OFFSET 1
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#define RTX_SSRC_OFFSET 2
#define RTP_CNAME "zlmediakit-rtp"
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#define RTP_LABEL "zlmediakit-label"
#define RTP_MSLABEL "zlmediakit-mslabel"
#define RTP_MSID RTP_MSLABEL " " RTP_LABEL
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using namespace std;
using namespace mediakit;
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// RTC配置项目
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namespace RTC {
#define RTC_FIELD "rtc."
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// rtp和rtcp接受超时时间
const string kTimeOutSec = RTC_FIELD "timeoutSec";
// 服务器外网ip
const string kExternIP = RTC_FIELD "externIP";
// 设置remb比特率非0时关闭twcc并开启remb。该设置在rtc推流时有效可以控制推流画质
const string kRembBitRate = RTC_FIELD "rembBitRate";
// webrtc单端口udp服务器
const string kPort = RTC_FIELD "port";
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static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
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mINI::Instance()[kRembBitRate] = 0;
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mINI::Instance()[kPort] = 8000;
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});
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} // namespace RTC
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static atomic<uint64_t> s_key { 0 };
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static void translateIPFromEnv(std::vector<std::string> &v) {
for (auto iter = v.begin(); iter != v.end();) {
if (start_with(*iter, "$")) {
auto ip = toolkit::getEnv(*iter);
if (ip.empty()) {
iter = v.erase(iter);
} else {
*iter++ = ip;
}
} else {
++iter;
}
}
}
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WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
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_poller = poller;
_identifier = "zlm_" + to_string(++s_key);
_packet_pool.setSize(64);
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}
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void WebRtcTransport::onCreate() {
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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, _identifier, makeRandStr(24));
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}
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void WebRtcTransport::onDestory() {
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#ifdef ENABLE_SCTP
_sctp = nullptr;
#endif
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_dtls_transport = nullptr;
_ice_server = nullptr;
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}
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const EventPoller::Ptr &WebRtcTransport::getPoller() const {
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return _poller;
}
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const string &WebRtcTransport::getIdentifier() const {
return _identifier;
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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) {
sendSockData((char *)packet->GetData(), packet->GetSize(), tuple);
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}
void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
InfoL;
}
void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) {
InfoL;
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}
void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) {
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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}
void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) {
InfoL;
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
void WebRtcTransport::OnDtlsTransportConnected(
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const RTC::DtlsTransport *dtlsTransport, RTC::SrtpSession::CryptoSuite srtpCryptoSuite, uint8_t *srtpLocalKey,
size_t srtpLocalKeyLen, uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, 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);
_srtp_session_recv = std::make_shared<RTC::SrtpSession>(
RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
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#ifdef ENABLE_SCTP
_sctp = std::make_shared<RTC::SctpAssociationImp>(getPoller(), this, 128, 128, 262144, true);
_sctp->TransportConnected();
#endif
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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) {
sendSockData((char *)data, len, nullptr);
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}
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void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
InfoL;
}
void WebRtcTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) {
InfoL;
onShutdown(SockException(Err_shutdown, "dtls transport failed"));
}
void WebRtcTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
InfoL;
onShutdown(SockException(Err_shutdown, "dtls close notify received"));
}
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void WebRtcTransport::OnDtlsTransportApplicationDataReceived(
const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
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#ifdef ENABLE_SCTP
_sctp->ProcessSctpData(data, len);
#else
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InfoL << hexdump(data, len);
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#endif
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
#ifdef ENABLE_SCTP
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void WebRtcTransport::OnSctpAssociationConnecting(RTC::SctpAssociation *sctpAssociation) {
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TraceL;
}
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void WebRtcTransport::OnSctpAssociationConnected(RTC::SctpAssociation *sctpAssociation) {
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InfoL << getIdentifier();
}
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void WebRtcTransport::OnSctpAssociationFailed(RTC::SctpAssociation *sctpAssociation) {
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WarnL << getIdentifier();
}
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void WebRtcTransport::OnSctpAssociationClosed(RTC::SctpAssociation *sctpAssociation) {
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InfoL << getIdentifier();
}
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void WebRtcTransport::OnSctpAssociationSendData(
RTC::SctpAssociation *sctpAssociation, const uint8_t *data, size_t len) {
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_dtls_transport->SendApplicationData(data, len);
}
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void WebRtcTransport::OnSctpAssociationMessageReceived(
RTC::SctpAssociation *sctpAssociation, uint16_t streamId, uint32_t ppid, const uint8_t *msg, size_t len) {
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InfoL << getIdentifier() << " " << streamId << " " << ppid << " " << len << " " << string((char *)msg, len);
RTC::SctpStreamParameters params;
params.streamId = streamId;
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// 回显数据
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_sctp->SendSctpMessage(params, ppid, msg, len);
}
#endif
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple) {
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auto pkt = _packet_pool.obtain2();
pkt->assign(buf, len);
onSendSockData(std::move(pkt), true, tuple ? tuple : _ice_server->GetSelectedTuple());
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}
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RTC::TransportTuple *WebRtcTransport::getSelectedTuple() const {
return _ice_server->GetSelectedTuple();
}
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void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
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auto remb = FCI_REMB::create({ ssrc }, (uint32_t)bit_rate);
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auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size());
fb->ssrc = htonl(0);
fb->ssrc_media = htonl(ssrc);
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sendRtcpPacket((char *)fb.get(), fb->getSize(), true);
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}
void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
pli->ssrc = htonl(0);
pli->ssrc_media = htonl(ssrc);
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sendRtcpPacket((char *)pli.get(), pli->getSize(), true);
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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);
for (auto &finger_prints : transport->GetLocalFingerprints()) {
if (finger_prints.algorithm == algorithm) {
return finger_prints.value;
}
}
throw std::invalid_argument(StrPrinter << "不支持的加密算法:" << algorithm_str);
}
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void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
// 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;
_dtls_transport->SetRemoteFingerprint(remote_fingerprint);
}
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void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
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// 开启remb后关闭twcc因为开启twcc后remb无效
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GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
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configure.enableTWCC(!remb_bit_rate);
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}
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std::string WebRtcTransport::getAnswerSdp(const string &offer) {
try {
//// 解析offer sdp ////
_offer_sdp = std::make_shared<RtcSession>();
_offer_sdp->loadFrom(offer);
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onCheckSdp(SdpType::offer, *_offer_sdp);
_offer_sdp->checkValid();
setRemoteDtlsFingerprint(*_offer_sdp);
//// sdp 配置 ////
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
RtcConfigure configure;
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configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
onRtcConfigure(configure);
//// 生成answer sdp ////
_answer_sdp = configure.createAnswer(*_offer_sdp);
onCheckSdp(SdpType::answer, *_answer_sdp);
_answer_sdp->checkValid();
return _answer_sdp->toString();
} catch (exception &ex) {
onShutdown(SockException(Err_shutdown, ex.what()));
throw;
}
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}
static bool is_dtls(char *buf) {
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return ((*buf > 19) && (*buf < 64));
}
static bool is_rtp(char *buf) {
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RtpHeader *header = (RtpHeader *)buf;
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return ((header->pt < 64) || (header->pt >= 96));
}
static 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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static string getPeerAddress(RTC::TransportTuple *tuple) {
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return SockUtil::inet_ntoa(tuple);
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}
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
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if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
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if (!packet) {
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WarnL << "parse stun error" << std::endl;
return;
}
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_ice_server->ProcessStunPacket(packet.get(), tuple);
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return;
}
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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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if (is_rtp(buf)) {
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if (!_srtp_session_recv) {
WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
return;
}
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if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
onRtp(buf, len, _ticker.createdTime());
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}
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return;
}
if (is_rtcp(buf)) {
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if (!_srtp_session_recv) {
WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
return;
}
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if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
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onRtcp(buf, len);
}
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return;
}
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}
void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
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if (_srtp_session_send) {
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auto pkt = _packet_pool.obtain2();
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// 预留rtx加入的两个字节
pkt->setCapacity((size_t)len + SRTP_MAX_TRAILER_LEN + 2);
pkt->assign(buf, len);
onBeforeEncryptRtp(pkt->data(), len, ctx);
if (_srtp_session_send->EncryptRtp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
pkt->setSize(len);
onSendSockData(std::move(pkt), flush);
}
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}
}
void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx) {
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if (_srtp_session_send) {
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auto pkt = _packet_pool.obtain2();
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// 预留rtx加入的两个字节
pkt->setCapacity((size_t)len + SRTP_MAX_TRAILER_LEN + 2);
pkt->assign(buf, len);
onBeforeEncryptRtcp(pkt->data(), len, ctx);
if (_srtp_session_send->EncryptRtcp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
pkt->setSize(len);
onSendSockData(std::move(pkt), flush);
}
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}
}
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///////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onCreate() {
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WebRtcTransport::onCreate();
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registerSelf();
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weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
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GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
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_timer = std::make_shared<Timer>(
timeoutSec / 2,
[weak_self]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
return false;
}
if (strong_self->_alive_ticker.elapsedTime() > timeoutSec * 1000) {
strong_self->onShutdown(SockException(Err_timeout, "接受rtp/rtcp/datachannel超时"));
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}
return true;
},
getPoller());
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_twcc_ctx.setOnSendTwccCB([this](uint32_t ssrc, string fci) { onSendTwcc(ssrc, fci); });
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}
void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
WebRtcTransport::OnDtlsTransportApplicationDataReceived(dtlsTransport, data, len);
#ifdef ENABLE_SCTP
if (_answer_sdp->isOnlyDatachannel()) {
_alive_ticker.resetTime();
}
#endif
}
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WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller)
: WebRtcTransport(poller) {
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InfoL << getIdentifier();
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}
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WebRtcTransportImp::~WebRtcTransportImp() {
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InfoL << getIdentifier();
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}
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void WebRtcTransportImp::onDestory() {
WebRtcTransport::onDestory();
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unregisterSelf();
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}
void WebRtcTransportImp::onSendSockData(Buffer::Ptr buf, bool flush, RTC::TransportTuple *tuple) {
if (!_selected_session) {
WarnL << "send data failed:" << buf->size();
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return;
}
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// 一次性发送一帧的rtp数据提高网络io性能
_selected_session->setSendFlushFlag(flush);
_selected_session->send(std::move(buf));
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}
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///////////////////////////////////////////////////////////////////
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bool WebRtcTransportImp::canSendRtp() const {
for (auto &m : _answer_sdp->media) {
if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::sendonly) {
return true;
}
}
return false;
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}
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bool WebRtcTransportImp::canRecvRtp() const {
for (auto &m : _answer_sdp->media) {
if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::recvonly) {
return true;
}
}
return false;
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}
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void WebRtcTransportImp::onStartWebRTC() {
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// 获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
for (auto &m_answer : _answer_sdp->media) {
if (m_answer.type == TrackApplication) {
continue;
}
auto m_offer = _offer_sdp->getMedia(m_answer.type);
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auto track = std::make_shared<MediaTrack>();
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track->media = &m_answer;
track->answer_ssrc_rtp = m_answer.getRtpSSRC();
track->answer_ssrc_rtx = m_answer.getRtxSSRC();
track->offer_ssrc_rtp = m_offer->getRtpSSRC();
track->offer_ssrc_rtx = m_offer->getRtxSSRC();
track->plan_rtp = &m_answer.plan[0];
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track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
track->rtcp_context_send = std::make_shared<RtcpContextForSend>();
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// rtp track type --> MediaTrack
if (m_answer.direction == RtpDirection::sendonly || m_answer.direction == RtpDirection::sendrecv) {
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// 该类型的track 才支持发送
_type_to_track[m_answer.type] = track;
}
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// send ssrc --> MediaTrack
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_ssrc_to_track[track->answer_ssrc_rtp] = track;
_ssrc_to_track[track->answer_ssrc_rtx] = track;
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// recv ssrc --> MediaTrack
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_ssrc_to_track[track->offer_ssrc_rtp] = track;
_ssrc_to_track[track->offer_ssrc_rtx] = track;
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// rtp pt --> MediaTrack
_pt_to_track.emplace(
track->plan_rtp->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtpTrack(track, _twcc_ctx, *this)));
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if (track->plan_rtx) {
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// rtx pt --> MediaTrack
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_pt_to_track.emplace(track->plan_rtx->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtxTrack(track)));
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}
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// 记录rtp ext类型与id的关系方便接收或发送rtp时修改rtp ext id
track->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
weak_ptr<MediaTrack> weak_track = track;
track->rtp_ext_ctx->setOnGetRtp([this, weak_track](uint8_t pt, uint32_t ssrc, const string &rid) {
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// ssrc --> MediaTrack
auto track = weak_track.lock();
assert(track);
_ssrc_to_track[ssrc] = std::move(track);
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InfoL << "get rtp, pt:" << (int)pt << ", ssrc:" << ssrc << ", rid:" << rid;
});
size_t index = 0;
for (auto &ssrc : m_offer->rtp_ssrc_sim) {
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// 记录ssrc对应的MediaTrack
_ssrc_to_track[ssrc.ssrc] = track;
if (m_offer->rtp_rids.size() > index) {
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// 支持firefox的simulcast, 提前映射好ssrc和rid的关系
track->rtp_ext_ctx->setRid(ssrc.ssrc, m_offer->rtp_rids[index]);
} else {
// SDP munging没有rid, 它通过group-ssrc:SIM给出ssrc列表;
// 系统又要有rid这里手工生成rid并为其绑定ssrc
std::string rid = "r" + std::to_string(index);
track->rtp_ext_ctx->setRid(ssrc.ssrc, rid);
if (ssrc.rtx_ssrc) {
track->rtp_ext_ctx->setRid(ssrc.rtx_ssrc, rid);
}
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}
++index;
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}
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}
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}
void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
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// 修改answer sdp的ip、端口信息
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str) {
std::vector<std::string> ret;
if (str.length()) {
ret = split(str, ",");
}
translateIPFromEnv(ret);
return ret;
});
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for (auto &m : sdp.media) {
m.addr.reset();
m.addr.address = extern_ips.empty() ? SockUtil::get_local_ip() : extern_ips[0];
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m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
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GET_CONFIG(uint16_t, local_port, RTC::kPort);
m.rtcp_addr.port = local_port;
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m.port = m.rtcp_addr.port;
sdp.origin.address = m.addr.address;
}
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if (!canSendRtp()) {
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// 设置我们发送的rtp的ssrc
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return;
}
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for (auto &m : sdp.media) {
if (m.type == TrackApplication) {
continue;
}
if (!m.rtp_rtx_ssrc.empty()) {
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// 已经生成了ssrc
continue;
}
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// 添加answer sdp的ssrc信息
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m.rtp_rtx_ssrc.emplace_back();
auto &ssrc = m.rtp_rtx_ssrc.back();
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// 发送的ssrc我们随便定义因为在发送rtp时会修改为此值
ssrc.ssrc = m.type + RTP_SSRC_OFFSET;
ssrc.cname = RTP_CNAME;
ssrc.label = RTP_LABEL;
ssrc.mslabel = RTP_MSLABEL;
ssrc.msid = RTP_MSID;
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if (m.getRelatedRtxPlan(m.plan[0].pt)) {
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// rtx ssrc
ssrc.rtx_ssrc = ssrc.ssrc + RTX_SSRC_OFFSET;
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}
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}
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}
void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) {
switch (type) {
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case SdpType::answer:
onCheckAnswer(sdp);
break;
case SdpType::offer:
break;
default: /*不可达*/
assert(0);
break;
}
}
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SdpAttrCandidate::Ptr
makeIceCandidate(std::string ip, uint16_t port, uint32_t priority = 100, std::string proto = "udp") {
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auto candidate = std::make_shared<SdpAttrCandidate>();
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// rtp端口
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candidate->component = 1;
candidate->transport = proto;
candidate->foundation = proto + "candidate";
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// 优先级单candidate时随便
candidate->priority = priority;
candidate->address = ip;
candidate->port = port;
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candidate->type = "host";
return candidate;
}
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
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GET_CONFIG(uint16_t, local_port, RTC::kPort);
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// 添加接收端口candidate信息
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str) {
std::vector<std::string> ret;
if (str.length()) {
ret = split(str, ",");
}
translateIPFromEnv(ret);
return ret;
});
if (extern_ips.empty()) {
std::string localIp = SockUtil::get_local_ip();
configure.addCandidate(*makeIceCandidate(localIp, local_port, 120, "udp"));
} else {
const uint32_t delta = 10;
uint32_t priority = 100 + delta * extern_ips.size();
for (auto ip : extern_ips) {
configure.addCandidate(*makeIceCandidate(ip, local_port, priority, "udp"));
priority -= delta;
}
}
}
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///////////////////////////////////////////////////////////////////
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class RtpChannel
: public RtpTrackImp
, public std::enable_shared_from_this<RtpChannel> {
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public:
RtpChannel(EventPoller::Ptr poller, RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) {
_poller = std::move(poller);
_on_nack = std::move(on_nack);
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setOnSorted(std::move(cb));
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_nack_ctx.setOnNack([this](const FCI_NACK &nack) { onNack(nack); });
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}
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~RtpChannel() override = default;
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RtpPacket::Ptr inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx) {
auto rtp = RtpTrack::inputRtp(type, sample_rate, ptr, len);
if (!rtp) {
return rtp;
}
auto seq = rtp->getSeq();
_nack_ctx.received(seq, is_rtx);
if (!is_rtx) {
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// 统计rtp接受情况便于生成nack rtcp包
_rtcp_context.onRtp(seq, rtp->getStamp(), rtp->ntp_stamp, sample_rate, len);
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}
return rtp;
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}
Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
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_rtcp_context.onRtcp(sr);
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return _rtcp_context.createRtcpRR(ssrc, getSSRC());
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}
float getLossRate() {
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auto expected = _rtcp_context.getExpectedPacketsInterval();
if (!expected) {
return -1;
}
return _rtcp_context.geLostInterval() * 100 / expected;
}
private:
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void starNackTimer() {
if (_delay_task) {
return;
}
weak_ptr<RtpChannel> weak_self = shared_from_this();
_delay_task = _poller->doDelayTask(10, [weak_self]() -> uint64_t {
auto strong_self = weak_self.lock();
if (!strong_self) {
return 0;
}
auto ret = strong_self->_nack_ctx.reSendNack();
if (!ret) {
strong_self->_delay_task = nullptr;
}
return ret;
});
}
void onNack(const FCI_NACK &nack) {
_on_nack(nack);
starNackTimer();
}
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private:
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NackContext _nack_ctx;
RtcpContextForRecv _rtcp_context;
EventPoller::Ptr _poller;
EventPoller::DelayTask::Ptr _delay_task;
function<void(const FCI_NACK &nack)> _on_nack;
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};
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std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const {
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auto it_chn = rtp_channel.find(rtp_ext_ctx->getRid(ssrc));
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if (it_chn == rtp_channel.end()) {
return nullptr;
}
return it_chn->second;
}
float WebRtcTransportImp::getLossRate(mediakit::TrackType type) {
for (auto &pr : _ssrc_to_track) {
auto ssrc = pr.first;
auto &track = pr.second;
auto rtp_chn = track->getRtpChannel(ssrc);
if (rtp_chn) {
if (track->media && type == track->media->type) {
return rtp_chn->getLossRate();
}
}
}
return -1;
}
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void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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_bytes_usage += 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) {
case RtcpType::RTCP_SR: {
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_alive_ticker.resetTime();
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// 对方汇报rtp发送情况
RtcpSR *sr = (RtcpSR *)rtcp;
auto it = _ssrc_to_track.find(sr->ssrc);
if (it != _ssrc_to_track.end()) {
auto &track = it->second;
auto rtp_chn = track->getRtpChannel(sr->ssrc);
if (!rtp_chn) {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
} else {
// InfoL << "接收丢包率,ssrc:" << sr->ssrc << ",loss rate(%):" << rtp_chn->getLossRate();
// 设置rtp时间戳与ntp时间戳的对应关系
rtp_chn->setNtpStamp(sr->rtpts, sr->getNtpUnixStampMS());
auto rr = rtp_chn->createRtcpRR(sr, track->answer_ssrc_rtp);
sendRtcpPacket(rr->data(), rr->size(), true);
}
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
}
break;
}
case RtcpType::RTCP_RR: {
_alive_ticker.resetTime();
// 对方汇报rtp接收情况
RtcpRR *rr = (RtcpRR *)rtcp;
for (auto item : rr->getItemList()) {
auto it = _ssrc_to_track.find(item->ssrc);
if (it != _ssrc_to_track.end()) {
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auto &track = it->second;
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track->rtcp_context_send->onRtcp(rtcp);
auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
sendRtcpPacket(sr->data(), sr->size(), true);
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} else {
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WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
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}
}
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break;
}
case RtcpType::RTCP_BYE: {
// 对方汇报停止发送rtp
RtcpBye *bye = (RtcpBye *)rtcp;
for (auto ssrc : bye->getSSRC()) {
auto it = _ssrc_to_track.find(*ssrc);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
continue;
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}
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_ssrc_to_track.erase(it);
}
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onRtcpBye();
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onShutdown(SockException(Err_eof, "rtcp bye message received"));
break;
}
case RtcpType::RTCP_PSFB:
case RtcpType::RTCP_RTPFB: {
if ((RtcpType)rtcp->pt == RtcpType::RTCP_PSFB) {
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break;
}
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// RTPFB
switch ((RTPFBType)rtcp->report_count) {
case RTPFBType::RTCP_RTPFB_NACK: {
RtcpFB *fb = (RtcpFB *)rtcp;
auto it = _ssrc_to_track.find(fb->ssrc_media);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
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}
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auto &track = it->second;
auto &fci = fb->getFci<FCI_NACK>();
track->nack_list.forEach(fci, [&](const RtpPacket::Ptr &rtp) {
// rtp重传
onSendRtp(rtp, true, true);
});
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break;
}
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default:
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break;
}
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break;
}
case RtcpType::RTCP_XR: {
RtcpXRRRTR *xr = (RtcpXRRRTR *)rtcp;
if (xr->bt != 4) {
break;
}
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auto it = _ssrc_to_track.find(xr->ssrc);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
auto &track = it->second;
track->rtcp_context_send->onRtcp(rtcp);
auto xrdlrr = track->rtcp_context_send->createRtcpXRDLRR(track->answer_ssrc_rtp, track->answer_ssrc_rtp);
sendRtcpPacket(xrdlrr->data(), xrdlrr->size(), true);
break;
}
default:
break;
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}
}
}
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///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track) {
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// rid --> RtpReceiverImp
auto &ref = track.rtp_channel[rid];
weak_ptr<WebRtcTransportImp> weak_self = dynamic_pointer_cast<WebRtcTransportImp>(shared_from_this());
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ref = std::make_shared<RtpChannel>(
getPoller(), [&track, this, rid](RtpPacket::Ptr rtp) mutable { onSortedRtp(track, rid, std::move(rtp)); },
[&track, weak_self, ssrc](const FCI_NACK &nack) mutable {
// nack发送可能由定时器异步触发
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->onSendNack(track, nack, ssrc);
}
});
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << track.plan_rtp->codec;
}
void WebRtcTransportImp::updateTicker() {
_alive_ticker.resetTime();
}
void WebRtcTransportImp::onRtp(const char *buf, size_t len, uint64_t stamp_ms) {
_bytes_usage += len;
_alive_ticker.resetTime();
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RtpHeader *rtp = (RtpHeader *)buf;
// 根据接收到的rtp的pt信息找到该流的信息
auto it = _pt_to_track.find(rtp->pt);
if (it == _pt_to_track.end()) {
WarnL << "unknown rtp pt:" << (int)rtp->pt;
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return;
}
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it->second->inputRtp(buf, len, stamp_ms, rtp);
}
void WrappedRtpTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
#if 0
auto seq = ntohs(rtp->seq);
if (track->media->type == TrackVideo && seq % 100 == 0) {
//此处模拟接受丢包
return;
}
#endif
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auto ssrc = ntohl(rtp->ssrc);
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// 修改ext id至统一
string rid;
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auto twcc_ext = track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
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if (twcc_ext) {
_twcc_ctx.onRtp(ssrc, twcc_ext.getTransportCCSeq(), stamp_ms);
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}
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auto &ref = track->rtp_channel[rid];
if (!ref) {
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_transport.createRtpChannel(rid, ssrc, *track);
}
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// 解析并排序rtp
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *)buf, len, false);
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}
void WrappedRtxTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
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// 修改ext id至统一
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string rid;
track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
auto &ref = track->rtp_channel[rid];
if (!ref) {
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// 再接收到对应的rtp前丢弃rtx包
WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ntohl(rtp->ssrc) << ", codec:" << track->plan_rtp->codec
<< ", seq:" << ntohs(rtp->seq);
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return;
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}
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// 这里是rtx重传包
// https://datatracker.ietf.org/doc/html/rfc4588#section-4
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auto payload = rtp->getPayloadData();
auto size = rtp->getPayloadSize(len);
if (size < 2) {
return;
}
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// 前两个字节是原始的rtp的seq
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auto origin_seq = payload[0] << 8 | payload[1];
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// rtx 转换为 rtp
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rtp->pt = track->plan_rtp->pt;
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rtp->seq = htons(origin_seq);
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rtp->ssrc = htonl(ref->getSSRC());
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memmove((uint8_t *)buf + 2, buf, payload - (uint8_t *)buf);
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buf += 2;
len -= 2;
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ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *)buf, len, true);
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}
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void WebRtcTransportImp::onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc) {
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auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
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rtcp->ssrc = htonl(track.answer_ssrc_rtp);
rtcp->ssrc_media = htonl(ssrc);
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sendRtcpPacket((char *)rtcp.get(), rtcp->getSize(), true);
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}
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void WebRtcTransportImp::onSendTwcc(uint32_t ssrc, const string &twcc_fci) {
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_TWCC, twcc_fci.data(), twcc_fci.size());
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rtcp->ssrc = htonl(0);
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rtcp->ssrc_media = htonl(ssrc);
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sendRtcpPacket((char *)rtcp.get(), rtcp->getSize(), true);
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
if (track.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
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// 定期发送pli请求关键帧方便非rtc等协议
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_pli_ticker.resetTime();
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sendRtcpPli(rtp->getSSRC());
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// 开启remb则发送remb包调节比特率
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GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
if (remb_bit_rate && _answer_sdp->supportRtcpFb(SdpConst::kRembRtcpFb)) {
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sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
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}
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}
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onRecvRtp(track, rid, std::move(rtp));
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx) {
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auto &track = _type_to_track[rtp->type];
if (!track) {
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// 忽略,对方不支持该编码类型
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return;
}
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if (!rtx) {
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// 统计rtp发送情况好做sr汇报
track->rtcp_context_send->onRtp(
rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate,
rtp->size() - RtpPacket::kRtpTcpHeaderSize);
track->nack_list.pushBack(rtp);
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#if 0
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//此处模拟发送丢包
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if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
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return;
}
#endif
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} else {
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// 发送rtx重传包
// TraceL << "send rtx rtp:" << rtp->getSeq();
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}
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pair<bool /*rtx*/, MediaTrack *> ctx { rtx, track.get() };
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sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
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_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
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}
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void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx) {
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auto pr = (pair<bool /*rtx*/, MediaTrack *> *)ctx;
auto header = (RtpHeader *)buf;
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if (!pr->first || !pr->second->plan_rtx) {
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// 普通的rtp,或者不支持rtx, 修改目标pt和ssrc
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pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
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header->pt = pr->second->plan_rtp->pt;
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header->ssrc = htonl(pr->second->answer_ssrc_rtp);
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} else {
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// 重传的rtp, rtx
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pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
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header->pt = pr->second->plan_rtx->pt;
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if (pr->second->answer_ssrc_rtx) {
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// 有rtx单独的ssrc,有些情况下浏览器支持rtx但是未指定rtx单独的ssrc
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header->ssrc = htonl(pr->second->answer_ssrc_rtx);
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} else {
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// 未单独指定rtx的ssrc那么使用rtp的ssrc
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header->ssrc = htonl(pr->second->answer_ssrc_rtp);
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}
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auto origin_seq = ntohs(header->seq);
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// seq跟原来的不一样
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header->seq = htons(_rtx_seq[pr->second->media->type]);
++_rtx_seq[pr->second->media->type];
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auto payload = header->getPayloadData();
auto payload_size = header->getPayloadSize(len);
if (payload_size) {
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// rtp负载后移两个字节这两个字节用于存放osn
// https://datatracker.ietf.org/doc/html/rfc4588#section-4
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memmove(payload + 2, payload, payload_size);
}
payload[0] = origin_seq >> 8;
payload[1] = origin_seq & 0xFF;
len += 2;
}
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}
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void WebRtcTransportImp::onShutdown(const SockException &ex) {
WarnL << ex.what();
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unrefSelf();
for (auto &pr : _history_sessions) {
auto session = pr.second.lock();
if (session) {
session->shutdown(ex);
}
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}
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}
void WebRtcTransportImp::setSession(Session::Ptr session) {
_history_sessions.emplace(session.get(), session);
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if (_selected_session) {
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InfoL << "rtc network changed: " << _selected_session->get_peer_ip() << ":"
<< _selected_session->get_peer_port() << " -> " << session->get_peer_ip() << ":"
<< session->get_peer_port() << ", id:" << getIdentifier();
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}
_selected_session = std::move(session);
unrefSelf();
}
const Session::Ptr &WebRtcTransportImp::getSession() const {
return _selected_session;
}
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uint64_t WebRtcTransportImp::getBytesUsage() const {
return _bytes_usage;
}
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uint64_t WebRtcTransportImp::getDuration() const {
return _alive_ticker.createdTime() / 1000;
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}
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void WebRtcTransportImp::onRtcpBye(){}
/////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::registerSelf() {
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_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
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WebRtcTransportManager::Instance().addItem(getIdentifier(), _self);
}
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void WebRtcTransportImp::unrefSelf() {
_self = nullptr;
}
void WebRtcTransportImp::unregisterSelf() {
unrefSelf();
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WebRtcTransportManager::Instance().removeItem(getIdentifier());
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}
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WebRtcTransportManager &WebRtcTransportManager::Instance() {
static WebRtcTransportManager s_instance;
return s_instance;
}
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void WebRtcTransportManager::addItem(const string &key, const WebRtcTransportImp::Ptr &ptr) {
lock_guard<mutex> lck(_mtx);
_map[key] = ptr;
}
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WebRtcTransportImp::Ptr WebRtcTransportManager::getItem(const string &key) {
if (key.empty()) {
return nullptr;
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}
lock_guard<mutex> lck(_mtx);
auto it = _map.find(key);
if (it == _map.end()) {
return nullptr;
}
return it->second.lock();
}
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void WebRtcTransportManager::removeItem(const string &key) {
lock_guard<mutex> lck(_mtx);
_map.erase(key);
}
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//////////////////////////////////////////////////////////////////////////////////////////////
WebRtcPluginManager &WebRtcPluginManager::Instance() {
static WebRtcPluginManager s_instance;
return s_instance;
}
void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
lock_guard<mutex> lck(_mtx_creator);
_map_creator[type] = std::move(cb);
}
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void WebRtcPluginManager::getAnswerSdp(
Session &sender, const string &type, const string &offer, const WebRtcArgs &args, const onCreateRtc &cb) {
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lock_guard<mutex> lck(_mtx_creator);
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
cb(WebRtcException(SockException(Err_other, "the type can not supported")));
return;
}
it->second(sender, offer, args, cb);
}
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#include "WebRtcEchoTest.h"
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#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
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void echo_plugin(
Session &sender, const string &offer, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
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void push_plugin(
Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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MediaInfo info(args["url"]);
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Broadcast::PublishAuthInvoker invoker = [cb, offer_sdp,
info](const string &err, const ProtocolOption &option) mutable {
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if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
}
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RtspMediaSourceImp::Ptr push_src;
std::shared_ptr<void> push_src_ownership;
auto src = MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid);
auto push_failed = (bool)src;
while (src) {
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// 尝试断连后继续推流
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auto rtsp_src = dynamic_pointer_cast<RtspMediaSourceImp>(src);
if (!rtsp_src) {
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// 源不是rtsp推流产生的
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break;
}
auto ownership = rtsp_src->getOwnership();
if (!ownership) {
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// 获取推流源所有权失败
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break;
}
push_src = std::move(rtsp_src);
push_src_ownership = std::move(ownership);
push_failed = false;
break;
}
if (push_failed) {
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cb(WebRtcException(SockException(Err_other, "already publishing")));
return;
}
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if (!push_src) {
push_src = std::make_shared<RtspMediaSourceImp>(info._vhost, info._app, info._streamid);
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
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}
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auto rtc
= WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
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push_src->setListener(rtc);
cb(*rtc);
};
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// rtsp推流需要鉴权
auto flag = NoticeCenter::Instance().emitEvent(
Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, static_cast<SockInfo &>(sender));
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if (!flag) {
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// 该事件无人监听,默认不鉴权
invoker("", ProtocolOption());
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}
}
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void play_plugin(
Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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MediaInfo info(args["url"]);
auto session_ptr = sender.shared_from_this();
Broadcast::AuthInvoker invoker = [cb, offer_sdp, info, session_ptr](const string &err) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
}
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// webrtc播放的是rtsp的源
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info._schema = RTSP_SCHEMA;
MediaSource::findAsync(info, session_ptr, [=](const MediaSource::Ptr &src_in) mutable {
auto src = dynamic_pointer_cast<RtspMediaSource>(src_in);
if (!src) {
cb(WebRtcException(SockException(Err_other, "stream not found")));
return;
}
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// 还原成rtc目的是为了hook时识别哪种播放协议
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info._schema = RTC_SCHEMA;
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
cb(*rtc);
});
};
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// 广播通用播放url鉴权事件
auto flag = NoticeCenter::Instance().emitEvent(
Broadcast::kBroadcastMediaPlayed, info, invoker, static_cast<SockInfo &>(sender));
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if (!flag) {
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// 该事件无人监听,默认不鉴权
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invoker("");
}
}
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static onceToken s_rtc_auto_register([]() {
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WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
});