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ZLMediaKit
Commit 02da99e2 by ziyue
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初步实现webrtc单udp端口模式

parent 8352f119
正在显示 包含 215 行增加 和 85 行删除
server/main.cpp
......@@ -25,7 +25,11 @@
#include "Rtp/RtpServer.h"
#include "WebApi.h"
#include "WebHook.h"
#include "../webrtc/Sdp.h"
#if defined(ENABLE_WEBRTC)
#include "../webrtc/WebRtcTransport.h"
#include "../webrtc/WebRtcSession.h"
#endif
#if defined(ENABLE_VERSION)
#include "Version.h"
......@@ -255,13 +259,13 @@ int start_main(int argc,char *argv[]) {
//加载配置文件，如果配置文件不存在就创建一个
loadIniConfig(g_ini_file.data());
if(!File::is_dir(ssl_file.data())){
if (!File::is_dir(ssl_file.data())) {
//不是文件夹，加载证书，证书包含公钥和私钥
SSL_Initor::Instance().loadCertificate(ssl_file.data());
}else{
} else {
//加载文件夹下的所有证书
File::scanDir(ssl_file,[](const string &path, bool isDir){
if(!isDir){
File::scanDir(ssl_file, [](const string &path, bool isDir) {
if (!isDir) {
//最后的一个证书会当做默认证书(客户端ssl握手时未指定主机)
SSL_Initor::Instance().loadCertificate(path.data());
}
......@@ -283,56 +287,67 @@ int start_main(int argc,char *argv[]) {
//简单的telnet服务器，可用于服务器调试，但是不能使用23端口，否则telnet上了莫名其妙的现象
//测试方法:telnet 127.0.0.1 9000
TcpServer::Ptr shellSrv(new TcpServer());
TcpServer::Ptr shellSrv = std::make_shared<TcpServer>();
//rtsp[s]服务器, 可用于诸如亚马逊echo show这样的设备访问
TcpServer::Ptr rtspSrv(new TcpServer());
TcpServer::Ptr rtspSSLSrv(new TcpServer());
TcpServer::Ptr rtspSrv = std::make_shared<TcpServer>();;
TcpServer::Ptr rtspSSLSrv = std::make_shared<TcpServer>();;
//rtmp[s]服务器
TcpServer::Ptr rtmpSrv(new TcpServer());
TcpServer::Ptr rtmpsSrv(new TcpServer());
TcpServer::Ptr rtmpSrv = std::make_shared<TcpServer>();;
TcpServer::Ptr rtmpsSrv = std::make_shared<TcpServer>();;
//http[s]服务器
TcpServer::Ptr httpSrv(new TcpServer());
TcpServer::Ptr httpsSrv(new TcpServer());
TcpServer::Ptr httpSrv = std::make_shared<TcpServer>();;
TcpServer::Ptr httpsSrv = std::make_shared<TcpServer>();;
#if defined(ENABLE_RTPPROXY)
//GB28181 rtp推流端口，支持UDP/TCP
RtpServer::Ptr rtpServer = std::make_shared<RtpServer>();
#endif//defined(ENABLE_RTPPROXY)
#if defined(ENABLE_WEBRTC)
//webrtc udp服务器
UdpServer::Ptr rtcSrv = std::make_shared<UdpServer>();
uint16_t rtcPort = mINI::Instance()[RTC::kPort];
#endif//defined(ENABLE_WEBRTC)
try {
//rtsp服务器，端口默认554
if(rtspPort) { rtspSrv->start<RtspSession>(rtspPort); }
if (rtspPort) { rtspSrv->start<RtspSession>(rtspPort); }
//rtsps服务器，端口默认322
if(rtspsPort) { rtspSSLSrv->start<RtspSessionWithSSL>(rtspsPort); }
if (rtspsPort) { rtspSSLSrv->start<RtspSessionWithSSL>(rtspsPort); }
//rtmp服务器，端口默认1935
if(rtmpPort) { rtmpSrv->start<RtmpSession>(rtmpPort); }
if (rtmpPort) { rtmpSrv->start<RtmpSession>(rtmpPort); }
//rtmps服务器，端口默认19350
if(rtmpsPort) { rtmpsSrv->start<RtmpSessionWithSSL>(rtmpsPort); }
if (rtmpsPort) { rtmpsSrv->start<RtmpSessionWithSSL>(rtmpsPort); }
//http服务器，端口默认80
if(httpPort) { httpSrv->start<HttpSession>(httpPort); }
if (httpPort) { httpSrv->start<HttpSession>(httpPort); }
//https服务器，端口默认443
if(httpsPort) { httpsSrv->start<HttpsSession>(httpsPort); }
if (httpsPort) { httpsSrv->start<HttpsSession>(httpsPort); }
//telnet远程调试服务器
if(shellPort) { shellSrv->start<ShellSession>(shellPort); }
if (shellPort) { shellSrv->start<ShellSession>(shellPort); }
#if defined(ENABLE_RTPPROXY)
//创建rtp服务器
if(rtpPort){ rtpServer->start(rtpPort); }
if (rtpPort) { rtpServer->start(rtpPort); }
#endif//defined(ENABLE_RTPPROXY)
}catch (std::exception &ex){
#if defined(ENABLE_WEBRTC)
//webrtc udp服务器
if (rtcPort) { rtcSrv->start<WebRtcSession>(rtcPort); }
#endif//defined(ENABLE_WEBRTC)
} catch (std::exception &ex) {
WarnL << "端口占用或无权限:" << ex.what() << endl;
ErrorL << "程序启动失败，请修改配置文件中端口号后重试!" << endl;
sleep(1);
#if !defined(_WIN32)
if(pid != getpid()){
kill(pid,SIGINT);
if (pid != getpid()) {
kill(pid, SIGINT);
}
#endif
return -1;
......
webrtc/WebRtcSession.cpp 0 → 100644
/*
* 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 "WebRtcSession.h"
#include "Util/util.h"
WebRtcSession::WebRtcSession(const Socket::Ptr &sock) : UdpSession(sock) {
socklen_t addr_len = sizeof(_peer_addr);
getpeername(sock->rawFD(), &_peer_addr, &addr_len);
InfoP(this);
}
WebRtcSession::~WebRtcSession() {
InfoP(this);
}
void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
auto buf = buffer->data();
auto len = buffer->size();
if (!_transport && 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;
}
if (packet->GetClass() == RTC::StunPacket::Class::REQUEST &&
packet->GetMethod() == RTC::StunPacket::Method::BINDING) {
//收到binding request请求
_transport = createTransport(packet->GetUsername());
}
}
if (_transport) {
_transport->inputSockData(buf, len, &_peer_addr);
}
}
void WebRtcSession::onError(const SockException &err) {
if (_transport) {
_transport->unrefSelf(err);
_transport = nullptr;
}
}
void WebRtcSession::onManager() {
}
std::shared_ptr<WebRtcTransport> WebRtcSession::createTransport(const string &user_name) {
if (user_name.empty()) {
return nullptr;
}
auto vec = split(user_name, ":");
auto ret = WebRtcTransportImp::getTransport(vec[0]);
ret->setSession(this);
return ret;
}
webrtc/WebRtcSession.h 0 → 100644
/*
* 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.
*/
#ifndef ZLMEDIAKIT_WEBRTCSESSION_H
#define ZLMEDIAKIT_WEBRTCSESSION_H
#include "Network/Session.h"
#include "IceServer.hpp"
#include "WebRtcTransport.h"
using namespace toolkit;
class WebRtcSession : public UdpSession {
public:
WebRtcSession(const Socket::Ptr &sock);
~WebRtcSession() override;
void onRecv(const Buffer::Ptr &) override;
void onError(const SockException &err) override;
void onManager() override;
private:
std::shared_ptr<WebRtcTransport> createTransport(const string &user_name);
private:
struct sockaddr _peer_addr;
std::shared_ptr<WebRtcTransport> _transport;
};
#endif //ZLMEDIAKIT_WEBRTCSESSION_H
webrtc/WebRtcTransport.cpp
......@@ -30,28 +30,59 @@ const string kTimeOutSec = RTC_FIELD"timeoutSec";
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";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
mINI::Instance()[kRembBitRate] = 0;
mINI::Instance()[kPort] = 8000;
});
}//namespace RTC
WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
_poller = poller;
_dtls_transport = std::make_shared<RTC::DtlsTransport>(poller, this);
_ice_server = std::make_shared<RTC::IceServer>(this, makeRandStr(4), makeRandStr(28).substr(4));
}
void WebRtcTransport::onCreate(){
_key = to_string(reinterpret_cast<uint64_t>(this));
_dtls_transport = std::make_shared<RTC::DtlsTransport>(_poller, this);
_ice_server = std::make_shared<RTC::IceServer>(this, _key, makeRandStr(24));
refSelf();
}
void WebRtcTransport::onDestory(){
_dtls_transport = nullptr;
_ice_server = nullptr;
unrefSelf(SockException());
}
static mutex s_rtc_mtx;
static unordered_map<string, weak_ptr<WebRtcTransportImp> > s_rtc_map;
void WebRtcTransport::refSelf() {
_self = shared_from_this();
lock_guard<mutex> lck(s_rtc_mtx);
s_rtc_map[_key] = static_pointer_cast<WebRtcTransportImp>(_self);
}
void WebRtcTransport::unrefSelf(const SockException &ex) {
_self = nullptr;
lock_guard<mutex> lck(s_rtc_mtx);
s_rtc_map.erase(_key);
}
WebRtcTransportImp::Ptr WebRtcTransportImp::getTransport(const string &key){
lock_guard<mutex> lck(s_rtc_mtx);
auto it = s_rtc_map.find(key);
if (it == s_rtc_map.end()) {
return nullptr;
}
return it->second.lock();
}
const EventPoller::Ptr& WebRtcTransport::getPoller() const{
......@@ -299,18 +330,7 @@ WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &polle
void WebRtcTransportImp::onCreate(){
WebRtcTransport::onCreate();
_socket = Socket::createSocket(getPoller(), false);
//随机端口，绑定全部网卡
_socket->bindUdpSock(0);
weak_ptr<WebRtcTransportImp> weak_self = shared_from_this();
_socket->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->inputSockData(buf->data(), buf->size(), addr);
}
});
_self = shared_from_this();
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();
......@@ -346,7 +366,7 @@ void WebRtcTransportImp::onDestory() {
<< _media_info._streamid
<< ")结束播放,耗时(s):" << duration;
if (_bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast<SockInfo &>(*_socket));
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, *static_cast<SockInfo *>(_session));
}
}
......@@ -357,7 +377,7 @@ void WebRtcTransportImp::onDestory() {
<< _media_info._streamid
<< ")结束推流,耗时(s):" << duration;
if (_bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast<SockInfo &>(*_socket));
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, *static_cast<SockInfo *>(_session));
}
}
}
......@@ -375,7 +395,7 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, const MediaInfo
void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
auto ptr = BufferRaw::create();
ptr->assign(buf, len);
_socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush);
_session->send(std::move(ptr));
}
///////////////////////////////////////////////////////////////////
......@@ -464,7 +484,7 @@ void WebRtcTransportImp::onStartWebRTC() {
}
if (canSendRtp()) {
_reader = _play_src->getRing()->attach(getPoller(), true);
weak_ptr<WebRtcTransportImp> weak_self = shared_from_this();
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
auto strongSelf = weak_self.lock();
if (!strongSelf) {
......@@ -516,7 +536,9 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
m.addr.address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
m.rtcp_addr.port = _socket->get_local_port();
GET_CONFIG(uint16_t, local_port, RTC::kPort);
m.rtcp_addr.port = local_port;
m.port = m.rtcp_addr.port;
sdp.origin.address = m.addr.address;
}
......@@ -576,7 +598,8 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
candidate->priority = 100;
GET_CONFIG(string, extern_ip, RTC::kExternIP);
candidate->address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
candidate->port = _socket->get_local_port();
GET_CONFIG(uint16_t, local_port, RTC::kPort);
candidate->port = local_port;
candidate->type = "host";
return candidate;
}
......@@ -871,7 +894,7 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPa
auto src_imp = std::make_shared<RtspMediaSourceImp>(_push_src->getVhost(), _push_src->getApp(), stream_id);
src_imp->setSdp(_push_src->getSdp());
src_imp->setProtocolTranslation(_push_src->isRecording(Recorder::type_hls),_push_src->isRecording(Recorder::type_mp4));
src_imp->setListener(shared_from_this());
src_imp->setListener(static_pointer_cast<WebRtcTransportImp>(shared_from_this()));
src = src_imp;
}
src->onWrite(std::move(rtp), false);
......@@ -943,7 +966,11 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx
void WebRtcTransportImp::onShutdown(const SockException &ex){
WarnL << ex.what();
_self = nullptr;
unrefSelf(ex);
if (_session) {
_session->shutdown(ex);
_session = nullptr;
}
}
/////////////////////////////////////////////////////////////////////////////////////////////
......@@ -975,27 +1002,9 @@ string WebRtcTransportImp::getOriginUrl(MediaSource &sender) const {
}
std::shared_ptr<SockInfo> WebRtcTransportImp::getOriginSock(MediaSource &sender) const {
return const_cast<WebRtcTransportImp *>(this)->shared_from_this();
}
/////////////////////////////////////////////////////////////////////////////////////////////
string WebRtcTransportImp::get_local_ip() {
return getSdp(SdpType::answer).media[0].candidate[0].address;
}
uint16_t WebRtcTransportImp::get_local_port() {
return _socket->get_local_port();
}
string WebRtcTransportImp::get_peer_ip() {
return SockUtil::inet_ntoa(((struct sockaddr_in *) getSelectedTuple())->sin_addr);
}
uint16_t WebRtcTransportImp::get_peer_port() {
return ntohs(((struct sockaddr_in *) getSelectedTuple())->sin_port);
return static_pointer_cast<SockInfo>(const_cast<Session *>(_session)->shared_from_this());
}
string WebRtcTransportImp::getIdentifier() const {
return StrPrinter << this;
void WebRtcTransportImp::setSession(Session *session) {
_session = session;
}
\ No newline at end of file
webrtc/WebRtcTransport.h
......@@ -23,16 +23,24 @@
#include "Rtcp/RtcpContext.h"
#include "Rtcp/RtcpFCI.h"
#include "Nack.h"
#include "Network/Session.h"
using namespace toolkit;
using namespace mediakit;
class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener {
//RTC配置项目
namespace RTC {
extern const string kPort;
}//namespace RTC
class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener, public std::enable_shared_from_this<WebRtcTransport> {
public:
using Ptr = std::shared_ptr<WebRtcTransport>;
WebRtcTransport(const EventPoller::Ptr &poller);
~WebRtcTransport() override = default;
void unrefSelf(const SockException &ex);
/**
* 创建对象
*/
......@@ -115,9 +123,11 @@ protected:
private:
void onSendSockData(const char *buf, size_t len, bool flush = true);
void setRemoteDtlsFingerprint(const RtcSession &remote);
void refSelf();
private:
uint8_t _srtp_buf[2000];
string _key;
EventPoller::Ptr _poller;
std::shared_ptr<RTC::IceServer> _ice_server;
std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
......@@ -125,6 +135,8 @@ private:
std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
RtcSession::Ptr _offer_sdp;
RtcSession::Ptr _answer_sdp;
//保持自我强引用
WebRtcTransport::Ptr _self;
};
class RtpChannel;
......@@ -149,7 +161,7 @@ public:
std::shared_ptr<RtpChannel> getRtpChannel(uint32_t ssrc) const;
};
class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public SockInfo, public std::enable_shared_from_this<WebRtcTransportImp>{
class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent{
public:
using Ptr = std::shared_ptr<WebRtcTransportImp>;
~WebRtcTransportImp() override;
......@@ -160,6 +172,9 @@ public:
* @return 对象
*/
static Ptr create(const EventPoller::Ptr &poller);
static Ptr getTransport(const string &key);
void setSession(Session *session);
/**
* 绑定rtsp媒体源
......@@ -193,18 +208,6 @@ protected:
// 获取媒体源客户端相关信息
std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
///////SockInfo override///////
//获取本机ip
string get_local_ip() override;
//获取本机端口号
uint16_t get_local_port() override;
//获取对方ip
string get_peer_ip() override;
//获取对方端口号
uint16_t get_peer_port() override;
//获取标识符
string getIdentifier() const override;
private:
WebRtcTransportImp(const EventPoller::Ptr &poller);
void onCreate() override;
......@@ -225,16 +228,14 @@ private:
uint64_t _bytes_usage = 0;
//媒体相关元数据
MediaInfo _media_info;
//保持自我强引用
Ptr _self;
//检测超时的定时器
Timer::Ptr _timer;
//刷新计时器
Ticker _alive_ticker;
//pli rtcp计时器
Ticker _pli_ticker;
//复合udp端口，接收一切rtp与rtcp
Socket::Ptr _socket;
//udp session
Session *_session;
//推流的rtsp源
RtspMediaSource::Ptr _push_src;
unordered_map<string/*rid*/, RtspMediaSource::Ptr> _push_src_simulcast;
......
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