ZLMediaKit/webrtc/WebRtcTransport.cpp
2024-05-12 18:15:11 +08:00

1467 lines
51 KiB
C++
Raw Blame History

This file contains ambiguous Unicode characters

This file contains Unicode characters that might be confused with other characters. If you think that this is intentional, you can safely ignore this warning. Use the Escape button to reveal them.

/*
* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
*
* Use of this source code is governed by MIT-like 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 <iostream>
#include <srtp2/srtp.h>
#include "Util/base64.h"
#include "Network/sockutil.h"
#include "Common/config.h"
#include "RtpExt.h"
#include "Rtcp/Rtcp.h"
#include "Rtcp/RtcpFCI.h"
#include "Rtcp/RtcpContext.h"
#include "Rtsp/Rtsp.h"
#include "Rtsp/RtpReceiver.h"
#include "WebRtcTransport.h"
#include "WebRtcEchoTest.h"
#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
#include "Rtsp/RtspMediaSourceImp.h"
#define RTP_SSRC_OFFSET 1
#define RTX_SSRC_OFFSET 2
#define RTP_CNAME "zlmediakit-rtp"
#define RTP_LABEL "zlmediakit-label"
#define RTP_MSLABEL "zlmediakit-mslabel"
using namespace std;
namespace mediakit {
// RTC配置项目
namespace Rtc {
#define RTC_FIELD "rtc."
// 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";
const string kTcpPort = RTC_FIELD "tcpPort";
// 比特率设置
const string kStartBitrate = RTC_FIELD "start_bitrate";
const string kMaxBitrate = RTC_FIELD "max_bitrate";
const string kMinBitrate = RTC_FIELD "min_bitrate";
// 数据通道设置
const string kDataChannelEcho = RTC_FIELD "datachannel_echo";
// rtp丢包状态最长保留时间
const string kNackMaxMS = RTC_FIELD "nackMaxMS";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
mINI::Instance()[kRembBitRate] = 0;
mINI::Instance()[kPort] = 8000;
mINI::Instance()[kTcpPort] = 8000;
mINI::Instance()[kStartBitrate] = 0;
mINI::Instance()[kMaxBitrate] = 0;
mINI::Instance()[kMinBitrate] = 0;
mINI::Instance()[kDataChannelEcho] = true;
mINI::Instance()[kNackMaxMS] = 3 * 1000;
});
} // namespace RTC
static atomic<uint64_t> s_key { 0 };
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;
}
}
}
static std::string getServerPrefix() {
//stun_user_name格式: base64(ip+udp_port+tcp_port) + _ + number
//其中ip为二进制char[4], udp_port/tcp_port为大端 uint16.
//number为自增长数确保短时间内唯一
GET_CONFIG(uint16_t, udp_port, Rtc::kPort);
GET_CONFIG(uint16_t, tcp_port, Rtc::kTcpPort);
char buf[8];
auto host = SockUtil::get_local_ip();
auto addr = SockUtil::make_sockaddr(host.data(), udp_port);
//拷贝ipv4地址
memcpy(buf, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_addr), 4);
//拷贝udp端口
memcpy(buf + 4, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_port), 2);
//tcp端口转大端模式
addr = SockUtil::make_sockaddr(host.data(), tcp_port);
//拷贝tcp端口
memcpy(buf + 6, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_port), 2);
auto ret = encodeBase64(string(buf, 8)) + '_';
InfoL << "MediaServer(" << host << ":" << udp_port << ":" << tcp_port << ") prefix: " << ret;
return ret;
}
const char* sockTypeStr(Session* session) {
if (session) {
switch (session->getSock()->sockType()) {
case SockNum::Sock_TCP: return "tcp";
case SockNum::Sock_UDP: return "udp";
default: break;
}
}
return "unknown";
}
WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
_poller = poller;
static auto prefix = getServerPrefix();
_identifier = prefix + to_string(++s_key);
_packet_pool.setSize(64);
}
void WebRtcTransport::onCreate() {
_dtls_transport = std::make_shared<RTC::DtlsTransport>(_poller, this);
_ice_server = std::make_shared<RTC::IceServer>(this, _identifier, makeRandStr(24));
}
void WebRtcTransport::onDestory() {
#ifdef ENABLE_SCTP
_sctp = nullptr;
#endif
_dtls_transport = nullptr;
_ice_server = nullptr;
}
const EventPoller::Ptr &WebRtcTransport::getPoller() const {
return _poller;
}
const string &WebRtcTransport::getIdentifier() const {
return _identifier;
}
const std::string& WebRtcTransport::deleteRandStr() const {
if (_delete_rand_str.empty()) {
_delete_rand_str = makeRandStr(32);
}
return _delete_rand_str;
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
void WebRtcTransport::OnIceServerSendStunPacket(
const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) {
sendSockData((char *)packet->GetData(), packet->GetSize(), tuple);
}
void WebRtcTransportImp::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
InfoL << getIdentifier() << " select tuple " << sockTypeStr(tuple) << " " << tuple->get_peer_ip() << ":" << tuple->get_peer_port();
tuple->setSendFlushFlag(false);
unrefSelf();
}
void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) {
InfoL << getIdentifier();
}
void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) {
InfoL << getIdentifier();
if (_answer_sdp->media[0].role == DtlsRole::passive) {
_dtls_transport->Run(RTC::DtlsTransport::Role::SERVER);
} else {
_dtls_transport->Run(RTC::DtlsTransport::Role::CLIENT);
}
}
void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) {
InfoL << getIdentifier();
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
void WebRtcTransport::OnDtlsTransportConnected(
const RTC::DtlsTransport *dtlsTransport, RTC::SrtpSession::CryptoSuite srtpCryptoSuite, uint8_t *srtpLocalKey,
size_t srtpLocalKeyLen, uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, std::string &remoteCert) {
InfoL << getIdentifier();
_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);
#ifdef ENABLE_SCTP
_sctp = std::make_shared<RTC::SctpAssociationImp>(getPoller(), this, 128, 128, 262144, true);
_sctp->TransportConnected();
#endif
onStartWebRTC();
}
#pragma pack(push, 1)
struct DtlsHeader {
uint8_t content_type;
uint16_t dtls_version;
uint16_t epoch;
uint8_t seq[6];
uint16_t length;
uint8_t payload[1];
};
#pragma pack(pop)
void WebRtcTransport::OnDtlsTransportSendData(
const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
size_t offset = 0;
while(offset < len) {
auto *header = reinterpret_cast<const DtlsHeader *>(data + offset);
auto length = ntohs(header->length) + offsetof(DtlsHeader, payload);
sendSockData((char *)data + offset, length, nullptr);
offset += length;
}
}
void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
InfoL << getIdentifier();
}
void WebRtcTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) {
InfoL << getIdentifier();
onShutdown(SockException(Err_shutdown, "dtls transport failed"));
}
void WebRtcTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
InfoL << getIdentifier();
onShutdown(SockException(Err_shutdown, "dtls close notify received"));
}
void WebRtcTransport::OnDtlsTransportApplicationDataReceived(
const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
#ifdef ENABLE_SCTP
_sctp->ProcessSctpData(data, len);
#else
InfoL << hexdump(data, len);
#endif
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
#ifdef ENABLE_SCTP
void WebRtcTransport::OnSctpAssociationConnecting(RTC::SctpAssociation *sctpAssociation) {
TraceL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpConnecting, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationConnected(RTC::SctpAssociation *sctpAssociation) {
InfoL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpConnected, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationFailed(RTC::SctpAssociation *sctpAssociation) {
WarnL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpFailed, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationClosed(RTC::SctpAssociation *sctpAssociation) {
InfoL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpClosed, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationSendData(
RTC::SctpAssociation *sctpAssociation, const uint8_t *data, size_t len) {
try {
NOTICE_EMIT(BroadcastRtcSctpSendArgs, Broadcast::kBroadcastRtcSctpSend, *this, data, len);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
_dtls_transport->SendApplicationData(data, len);
}
void WebRtcTransport::OnSctpAssociationMessageReceived(
RTC::SctpAssociation *sctpAssociation, uint16_t streamId, uint32_t ppid, const uint8_t *msg, size_t len) {
InfoL << getIdentifier() << " " << streamId << " " << ppid << " " << len << " " << string((char *)msg, len);
RTC::SctpStreamParameters params;
params.streamId = streamId;
GET_CONFIG(bool, datachannel_echo, Rtc::kDataChannelEcho);
if (datachannel_echo) {
// 回显数据
_sctp->SendSctpMessage(params, ppid, msg, len);
}
try {
NOTICE_EMIT(BroadcastRtcSctpReceivedArgs, Broadcast::kBroadcastRtcSctpReceived, *this, streamId, ppid, msg, len);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
#endif
void WebRtcTransport::sendDatachannel(uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
#ifdef ENABLE_SCTP
if (_sctp) {
RTC::SctpStreamParameters params;
params.streamId = streamId;
_sctp->SendSctpMessage(params, ppid, (uint8_t *)msg, len);
}
#else
WarnL << "WebRTC datachannel disabled!";
#endif
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple) {
auto pkt = _packet_pool.obtain2();
pkt->assign(buf, len);
onSendSockData(std::move(pkt), true, tuple ? tuple : _ice_server->GetSelectedTuple());
}
Session::Ptr WebRtcTransport::getSession() const {
auto tuple = _ice_server ? _ice_server->GetSelectedTuple(true) : nullptr;
return tuple ? static_pointer_cast<Session>(tuple->shared_from_this()) : nullptr;
}
void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
auto remb = FCI_REMB::create({ ssrc }, (uint32_t)bit_rate);
auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size());
fb->ssrc = htonl(0);
fb->ssrc_media = htonl(ssrc);
sendRtcpPacket((char *)fb.get(), fb->getSize(), true);
}
void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
pli->ssrc = htonl(0);
pli->ssrc_media = htonl(ssrc);
sendRtcpPacket((char *)pli.get(), pli->getSize(), true);
}
string getFingerprint(const string &algorithm_str, const std::shared_ptr<RTC::DtlsTransport> &transport) {
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);
}
void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
// 设置远端dtls签名
RTC::DtlsTransport::Fingerprint remote_fingerprint;
remote_fingerprint.algorithm
= RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm);
remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash;
_dtls_transport->SetRemoteFingerprint(remote_fingerprint);
}
void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
// 开启remb后关闭twcc因为开启twcc后remb无效
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
configure.enableTWCC(!remb_bit_rate);
}
static void setSdpBitrate(RtcSession &sdp) {
GET_CONFIG(size_t, max_bitrate, Rtc::kMaxBitrate);
GET_CONFIG(size_t, min_bitrate, Rtc::kMinBitrate);
GET_CONFIG(size_t, start_bitrate, Rtc::kStartBitrate);
auto m = (RtcMedia *)(sdp.getMedia(TrackType::TrackVideo));
if (m) {
auto &plan = m->plan[0];
if (max_bitrate) plan.fmtp["x-google-max-bitrate"] = std::to_string(max_bitrate);
if (min_bitrate) plan.fmtp["x-google-min-bitrate"] = std::to_string(min_bitrate);
if (start_bitrate) plan.fmtp["x-google-start-bitrate"] = std::to_string(start_bitrate);
}
}
std::string WebRtcTransport::getAnswerSdp(const string &offer) {
try {
//// 解析offer sdp ////
_offer_sdp = std::make_shared<RtcSession>();
_offer_sdp->loadFrom(offer);
onCheckSdp(SdpType::offer, *_offer_sdp);
_offer_sdp->checkValid();
setRemoteDtlsFingerprint(*_offer_sdp);
//// sdp 配置 ////
RtcConfigure configure;
onRtcConfigure(configure);
//// 生成answer sdp ////
_answer_sdp = configure.createAnswer(*_offer_sdp);
onCheckSdp(SdpType::answer, *_answer_sdp);
setSdpBitrate(*_answer_sdp);
_answer_sdp->checkValid();
return _answer_sdp->toString();
} catch (exception &ex) {
onShutdown(SockException(Err_shutdown, ex.what()));
throw;
}
}
static bool isDtls(char *buf) {
return ((*buf > 19) && (*buf < 64));
}
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
if (!packet) {
WarnL << "parse stun error";
return;
}
_ice_server->ProcessStunPacket(packet.get(), tuple);
return;
}
if (isDtls(buf)) {
_dtls_transport->ProcessDtlsData((uint8_t *)buf, len);
return;
}
if (isRtp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
onRtp(buf, len, _ticker.createdTime());
}
return;
}
if (isRtcp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtcp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
onRtcp(buf, len);
}
return;
}
}
void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
if (_srtp_session_send) {
auto pkt = _packet_pool.obtain2();
// 预留rtx加入的两个字节
pkt->setCapacity((size_t)len + SRTP_MAX_TRAILER_LEN + 2);
memcpy(pkt->data(), 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);
}
}
}
void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx) {
if (_srtp_session_send) {
auto pkt = _packet_pool.obtain2();
// 预留rtx加入的两个字节
pkt->setCapacity((size_t)len + SRTP_MAX_TRAILER_LEN + 2);
memcpy(pkt->data(), 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);
}
}
}
///////////////////////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onCreate() {
WebRtcTransport::onCreate();
registerSelf();
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
GET_CONFIG(float, timeoutSec, Rtc::kTimeOutSec);
_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"));
}
return true;
},
getPoller());
_twcc_ctx.setOnSendTwccCB([this](uint32_t ssrc, string fci) { onSendTwcc(ssrc, fci); });
}
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
}
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
InfoL << getIdentifier();
}
WebRtcTransportImp::~WebRtcTransportImp() {
InfoL << getIdentifier();
}
void WebRtcTransportImp::onDestory() {
WebRtcTransport::onDestory();
unregisterSelf();
}
void WebRtcTransportImp::onSendSockData(Buffer::Ptr buf, bool flush, RTC::TransportTuple *tuple) {
if (tuple == nullptr) {
tuple = _ice_server->GetSelectedTuple();
if (!tuple) {
WarnL << "send data failed:" << buf->size();
return;
}
}
// 一次性发送一帧的rtp数据提高网络io性能
if (tuple->getSock()->sockType() == SockNum::Sock_TCP) {
// 增加tcp两字节头
auto len = buf->size();
char tcp_len[2] = { 0 };
tcp_len[0] = (len >> 8) & 0xff;
tcp_len[1] = len & 0xff;
tuple->SockSender::send(tcp_len, 2);
}
tuple->send(std::move(buf));
if (flush) {
tuple->flushAll();
}
}
///////////////////////////////////////////////////////////////////
bool WebRtcTransportImp::canSendRtp() const {
for (auto &m : _answer_sdp->media) {
if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::sendonly) {
return true;
}
}
return false;
}
bool WebRtcTransportImp::canRecvRtp() const {
for (auto &m : _answer_sdp->media) {
if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::recvonly) {
return true;
}
}
return false;
}
void WebRtcTransportImp::onStartWebRTC() {
// 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);
auto track = std::make_shared<MediaTrack>();
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];
track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
track->rtcp_context_send = std::make_shared<RtcpContextForSend>();
// rtp track type --> MediaTrack
if (m_answer.direction == RtpDirection::sendonly || m_answer.direction == RtpDirection::sendrecv) {
// 该类型的track 才支持发送
_type_to_track[m_answer.type] = track;
}
// send ssrc --> MediaTrack
_ssrc_to_track[track->answer_ssrc_rtp] = track;
_ssrc_to_track[track->answer_ssrc_rtx] = track;
// recv ssrc --> MediaTrack
_ssrc_to_track[track->offer_ssrc_rtp] = track;
_ssrc_to_track[track->offer_ssrc_rtx] = track;
// rtp pt --> MediaTrack
_pt_to_track.emplace(
track->plan_rtp->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtpTrack(track, _twcc_ctx, *this)));
if (track->plan_rtx) {
// rtx pt --> MediaTrack
_pt_to_track.emplace(track->plan_rtx->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtxTrack(track)));
}
// 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_answer);
weak_ptr<MediaTrack> weak_track = track;
track->rtp_ext_ctx->setOnGetRtp([this, weak_track](uint8_t pt, uint32_t ssrc, const string &rid) {
// ssrc --> MediaTrack
auto track = weak_track.lock();
assert(track);
_ssrc_to_track[ssrc] = std::move(track);
InfoL << "get rtp, pt:" << (int)pt << ", ssrc:" << ssrc << ", rid:" << rid;
});
size_t index = 0;
for (auto &ssrc : m_offer->rtp_ssrc_sim) {
// 记录ssrc对应的MediaTrack
_ssrc_to_track[ssrc.ssrc] = track;
if (m_offer->rtp_rids.size() > index) {
// 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);
}
}
++index;
}
}
}
void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
// 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;
});
for (auto &m : sdp.media) {
m.addr.reset();
m.addr.address = extern_ips.empty() ? _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip : extern_ips[0];
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
GET_CONFIG(uint16_t, udp_port, Rtc::kPort);
GET_CONFIG(uint16_t, tcp_port, Rtc::kTcpPort);
m.port = m.port ? (udp_port ? udp_port : tcp_port) : 0;
if (m.type != TrackApplication) {
m.rtcp_addr.port = m.port;
}
sdp.origin.address = m.addr.address;
}
if (!canSendRtp()) {
// 设置我们发送的rtp的ssrc
return;
}
for (auto &m : sdp.media) {
if (m.type == TrackApplication) {
continue;
}
if (!m.rtp_rtx_ssrc.empty()) {
// 已经生成了ssrc
continue;
}
// 添加answer sdp的ssrc信息
m.rtp_rtx_ssrc.emplace_back();
auto &ssrc = m.rtp_rtx_ssrc.back();
// 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 = std::string(RTP_LABEL) + '-' + m.mid;
ssrc.mslabel = RTP_MSLABEL;
ssrc.msid = ssrc.mslabel + ' ' + ssrc.label;
if (m.getRelatedRtxPlan(m.plan[0].pt)) {
// rtx ssrc
ssrc.rtx_ssrc = ssrc.ssrc + RTX_SSRC_OFFSET;
}
}
}
void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) {
switch (type) {
case SdpType::answer:
onCheckAnswer(sdp);
break;
case SdpType::offer:
break;
default: /*不可达*/
assert(0);
break;
}
}
SdpAttrCandidate::Ptr
makeIceCandidate(std::string ip, uint16_t port, uint32_t priority = 100, std::string proto = "udp") {
auto candidate = std::make_shared<SdpAttrCandidate>();
// rtp端口
candidate->component = 1;
candidate->transport = proto;
candidate->foundation = proto + "candidate";
// 优先级单candidate时随便
candidate->priority = priority;
candidate->address = std::move(ip);
candidate->port = port;
candidate->type = "host";
if (proto == "tcp") {
candidate->type += " tcptype passive";
}
return candidate;
}
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
if (!_cands.empty()) {
for (auto &cand : _cands) {
configure.addCandidate(cand);
}
return;
}
GET_CONFIG(uint16_t, local_udp_port, Rtc::kPort);
GET_CONFIG(uint16_t, local_tcp_port, Rtc::kTcpPort);
// 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 local_ip = _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip;
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
} else {
const uint32_t delta = 10;
uint32_t priority = 100 + delta * extern_ips.size();
for (auto ip : extern_ips) {
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(ip, local_udp_port, priority, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority - (_preferred_tcp ? -5 : 5), "tcp")); }
priority -= delta;
}
}
}
void WebRtcTransportImp::setPreferredTcp(bool flag) {
_preferred_tcp = flag;
}
void WebRtcTransportImp::setLocalIp(std::string local_ip) {
_local_ip = std::move(local_ip);
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
}
///////////////////////////////////////////////////////////////////
class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {
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);
setOnSorted(std::move(cb));
//设置jitter buffer参数
GET_CONFIG(uint32_t, nack_maxms, Rtc::kNackMaxMS);
RtpTrackImp::setParams(1024, nack_maxms, 512);
_nack_ctx.setOnNack([this](const FCI_NACK &nack) { onNack(nack); });
}
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) {
// 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);
}
return rtp;
}
Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
_rtcp_context.onRtcp(sr);
return _rtcp_context.createRtcpRR(ssrc, getSSRC());
}
float getLossRate() {
auto expected = _rtcp_context.getExpectedPacketsInterval();
if (!expected) {
return -1;
}
return _rtcp_context.getLostInterval() * 100 / expected;
}
private:
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();
}
private:
NackContext _nack_ctx;
RtcpContextForRecv _rtcp_context;
EventPoller::Ptr _poller;
EventPoller::DelayTask::Ptr _delay_task;
function<void(const FCI_NACK &nack)> _on_nack;
};
std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const {
auto it_chn = rtp_channel.find(rtp_ext_ctx->getRid(ssrc));
if (it_chn == rtp_channel.end()) {
return nullptr;
}
return it_chn->second;
}
float WebRtcTransportImp::getLossRate(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;
}
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
_bytes_usage += len;
auto rtcps = RtcpHeader::loadFromBytes((char *)buf, len);
for (auto rtcp : rtcps) {
switch ((RtcpType)rtcp->pt) {
case RtcpType::RTCP_SR: {
_alive_ticker.resetTime();
// 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 {
// 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()) {
auto &track = it->second;
track->rtcp_context_send->onRtcp(rtcp);
auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
sendRtcpPacket(sr->data(), sr->size(), true);
} else {
WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
}
}
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;
}
_ssrc_to_track.erase(it);
}
onRtcpBye();
// bye 会在 sender audio track mute 时出现, 因此不能作为 shutdown 的依据
break;
}
case RtcpType::RTCP_PSFB:
case RtcpType::RTCP_RTPFB: {
if ((RtcpType)rtcp->pt == RtcpType::RTCP_PSFB) {
break;
}
// 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;
}
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);
});
break;
}
default:
break;
}
break;
}
case RtcpType::RTCP_XR: {
RtcpXRRRTR *xr = (RtcpXRRRTR *)rtcp;
if (xr->bt != 4) {
break;
}
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;
}
}
}
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track) {
// rid --> RtpReceiverImp
auto &ref = track.rtp_channel[rid];
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
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();
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;
return;
}
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
auto ssrc = ntohl(rtp->ssrc);
// 修改ext id至统一
string rid;
auto twcc_ext = track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
if (twcc_ext) {
_twcc_ctx.onRtp(ssrc, twcc_ext.getTransportCCSeq(), stamp_ms);
}
auto &ref = track->rtp_channel[rid];
if (!ref) {
_transport.createRtpChannel(rid, ssrc, *track);
}
// 解析并排序rtp
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *)buf, len, false);
}
void WrappedRtxTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
// 修改ext id至统一
string rid;
track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
auto &ref = track->rtp_channel[rid];
if (!ref) {
// 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);
return;
}
// 这里是rtx重传包
// https://datatracker.ietf.org/doc/html/rfc4588#section-4
auto payload = rtp->getPayloadData();
auto size = rtp->getPayloadSize(len);
if (size < 2) {
return;
}
// 前两个字节是原始的rtp的seq
auto origin_seq = payload[0] << 8 | payload[1];
// rtx 转换为 rtp
rtp->pt = track->plan_rtp->pt;
rtp->seq = htons(origin_seq);
rtp->ssrc = htonl(ref->getSSRC());
memmove((uint8_t *)buf + 2, buf, payload - (uint8_t *)buf);
buf += 2;
len -= 2;
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *)buf, len, true);
}
void WebRtcTransportImp::onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc) {
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
rtcp->ssrc = htonl(track.answer_ssrc_rtp);
rtcp->ssrc_media = htonl(ssrc);
sendRtcpPacket((char *)rtcp.get(), rtcp->getSize(), true);
}
void WebRtcTransportImp::onSendTwcc(uint32_t ssrc, const string &twcc_fci) {
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_TWCC, twcc_fci.data(), twcc_fci.size());
rtcp->ssrc = htonl(0);
rtcp->ssrc_media = htonl(ssrc);
sendRtcpPacket((char *)rtcp.get(), rtcp->getSize(), true);
}
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
if (track.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
// 定期发送pli请求关键帧方便非rtc等协议
_pli_ticker.resetTime();
sendRtcpPli(rtp->getSSRC());
// 开启remb则发送remb包调节比特率
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
if (remb_bit_rate && _answer_sdp->supportRtcpFb(SdpConst::kRembRtcpFb)) {
sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
}
}
onRecvRtp(track, rid, std::move(rtp));
}
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx) {
auto &track = _type_to_track[rtp->type];
if (!track) {
// 忽略,对方不支持该编码类型
return;
}
if (!rtx) {
// 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);
#if 0
//此处模拟发送丢包
if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
return;
}
#endif
} else {
// 发送rtx重传包
// TraceL << "send rtx rtp:" << rtp->getSeq();
}
pair<bool /*rtx*/, MediaTrack *> ctx { rtx, track.get() };
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
}
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx) {
auto pr = (pair<bool /*rtx*/, MediaTrack *> *)ctx;
auto header = (RtpHeader *)buf;
if (!pr->first || !pr->second->plan_rtx) {
// 普通的rtp,或者不支持rtx, 修改目标pt和ssrc
pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
header->pt = pr->second->plan_rtp->pt;
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
} else {
// 重传的rtp, rtx
pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
header->pt = pr->second->plan_rtx->pt;
if (pr->second->answer_ssrc_rtx) {
// 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
header->ssrc = htonl(pr->second->answer_ssrc_rtx);
} else {
// 未单独指定rtx的ssrc那么使用rtp的ssrc
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
}
auto origin_seq = ntohs(header->seq);
// seq跟原来的不一样
header->seq = htons(_rtx_seq[pr->second->media->type]);
++_rtx_seq[pr->second->media->type];
auto payload = header->getPayloadData();
auto payload_size = header->getPayloadSize(len);
if (payload_size) {
// 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
memmove(payload + 2, payload, payload_size);
}
payload[0] = origin_seq >> 8;
payload[1] = origin_seq & 0xFF;
len += 2;
}
}
void WebRtcTransportImp::safeShutdown(const SockException &ex) {
std::weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
getPoller()->async([ex, weak_self]() {
if (auto strong_self = weak_self.lock()) {
strong_self->onShutdown(ex);
}
});
}
void WebRtcTransportImp::onShutdown(const SockException &ex) {
WarnL << ex;
unrefSelf();
for (auto &tuple : _ice_server->GetTuples()) {
tuple->shutdown(ex);
}
}
void WebRtcTransportImp::removeTuple(RTC::TransportTuple *tuple) {
InfoL << getIdentifier() << " remove tuple " << tuple->get_peer_ip() << ":" << tuple->get_peer_port();
this->_ice_server->RemoveTuple(tuple);
}
uint64_t WebRtcTransportImp::getBytesUsage() const {
return _bytes_usage;
}
uint64_t WebRtcTransportImp::getDuration() const {
return _alive_ticker.createdTime() / 1000;
}
void WebRtcTransportImp::onRtcpBye(){}
/////////////////////////////////////////////////////////////////////////////////////////////
void WebRtcTransportImp::registerSelf() {
_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
WebRtcTransportManager::Instance().addItem(getIdentifier(), _self);
}
void WebRtcTransportImp::unrefSelf() {
_self = nullptr;
}
void WebRtcTransportImp::unregisterSelf() {
unrefSelf();
WebRtcTransportManager::Instance().removeItem(getIdentifier());
}
WebRtcTransportManager &WebRtcTransportManager::Instance() {
static WebRtcTransportManager s_instance;
return s_instance;
}
void WebRtcTransportManager::addItem(const string &key, const WebRtcTransportImp::Ptr &ptr) {
lock_guard<mutex> lck(_mtx);
_map[key] = ptr;
}
WebRtcTransportImp::Ptr WebRtcTransportManager::getItem(const string &key) {
if (key.empty()) {
return nullptr;
}
lock_guard<mutex> lck(_mtx);
auto it = _map.find(key);
if (it == _map.end()) {
return nullptr;
}
return it->second.lock();
}
void WebRtcTransportManager::removeItem(const string &key) {
lock_guard<mutex> lck(_mtx);
_map.erase(key);
}
//////////////////////////////////////////////////////////////////////////////////////////////
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);
}
void WebRtcPluginManager::setListener(Listener cb) {
lock_guard<mutex> lck(_mtx_creator);
_listener = std::move(cb);
}
void WebRtcPluginManager::negotiateSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateWebRtc &cb_in) {
onCreateWebRtc cb;
lock_guard<mutex> lck(_mtx_creator);
if (_listener) {
auto listener = _listener;
auto args_ptr = args.shared_from_this();
auto sender_ptr = static_pointer_cast<Session>(sender.shared_from_this());
cb = [listener, sender_ptr, type, args_ptr, cb_in](const WebRtcInterface &rtc) {
listener(*sender_ptr, type, *args_ptr, rtc);
cb_in(rtc);
};
} else {
cb = cb_in;
}
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
cb_in(WebRtcException(SockException(Err_other, "the type can not supported")));
return;
}
it->second(sender, args, cb);
}
void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
}
RtspMediaSourceImp::Ptr push_src;
std::shared_ptr<void> push_src_ownership;
auto src = MediaSource::find(RTSP_SCHEMA, info.vhost, info.app, info.stream);
auto push_failed = (bool)src;
while (src) {
// 尝试断连后继续推流
auto rtsp_src = dynamic_pointer_cast<RtspMediaSourceImp>(src);
if (!rtsp_src) {
// 源不是rtsp推流产生的
break;
}
auto ownership = rtsp_src->getOwnership();
if (!ownership) {
// 获取推流源所有权失败
break;
}
push_src = std::move(rtsp_src);
push_src_ownership = std::move(ownership);
push_failed = false;
break;
}
if (push_failed) {
cb(WebRtcException(SockException(Err_other, "already publishing")));
return;
}
if (!push_src) {
push_src = std::make_shared<RtspMediaSourceImp>(info);
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
}
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
push_src->setListener(rtc);
cb(*rtc);
};
// rtsp推流需要鉴权
auto flag = NOTICE_EMIT(BroadcastMediaPublishArgs, Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, sender);
if (!flag) {
// 该事件无人监听,默认不鉴权
invoker("", ProtocolOption());
}
}
void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
auto session_ptr = static_pointer_cast<Session>(sender.shared_from_this());
Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
}
// webrtc播放的是rtsp的源
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;
}
// 还原成rtc目的是为了hook时识别哪种播放协议
info.schema = "rtc";
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
cb(*rtc);
});
};
// 广播通用播放url鉴权事件
auto flag = NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, info, invoker, sender);
if (!flag) {
// 该事件无人监听,默认不鉴权
invoker("");
}
}
static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
{
static auto is_vaild_ip = [](const std::string &ip) -> bool {
int a, b, c, d;
return sscanf(ip.c_str(), "%d.%d.%d.%d", &a, &b, &c, &d) == 4;
};
std::string host = args["Host"];
if (!host.empty()) {
auto local_ip = host.substr(0, host.find(':'));
if (!is_vaild_ip(local_ip) || local_ip == "127.0.0.1") {
local_ip = "";
}
rtc.setLocalIp(std::move(local_ip));
}
}
bool preferred_tcp = args["preferred_tcp"];
{
rtc.setPreferredTcp(preferred_tcp);
}
{
vector<SdpAttrCandidate> cands;
{
auto cand_str = trim(args["cand_udp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// udp优先
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
cands.emplace_back(std::move(*ice_cand));
}
}
{
auto cand_str = trim(args["cand_tcp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// tcp模式
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 120 : 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
rtc.setIceCandidate(std::move(cands));
}
}
}
static onceToken s_rtc_auto_register([]() {
#if !defined (NDEBUG)
// debug模式才开启echo插件
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
#endif
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
WebRtcPluginManager::Instance().setListener([](Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
});
});
}// namespace mediakit