初步实现webrtc单udp端口模式

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){
+            File::scanDir(ssl_file, [](const string &path, bool isDir) {
-                if(!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 rtspSrv = std::make_shared<TcpServer>();;
-        TcpServer::Ptr rtspSSLSrv(new TcpServer());
+        TcpServer::Ptr rtspSSLSrv = std::make_shared<TcpServer>();;
 
         //rtmp[s]服务器
-        TcpServer::Ptr rtmpSrv(new TcpServer());
+        TcpServer::Ptr rtmpSrv = std::make_shared<TcpServer>();;
-        TcpServer::Ptr rtmpsSrv(new TcpServer());
+        TcpServer::Ptr rtmpsSrv = std::make_shared<TcpServer>();;
 
         //http[s]服务器
-        TcpServer::Ptr httpSrv(new TcpServer());
+        TcpServer::Ptr httpSrv = std::make_shared<TcpServer>();;
-        TcpServer::Ptr httpsSrv(new 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()){
+            if (pid != getpid()) {
-                kill(pid,SIGINT);
+                kill(pid, SIGINT);
             }
 #endif
             return -1;

webrtc/WebRtcSession.cpp

@@ -0,0 +1,65 @@
+/*
+ * 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

@@ -0,0 +1,39 @@
+/*
+ * 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);
+    weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
-    //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();
-
     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();
+    return static_pointer_cast<SockInfo>(const_cast<Session *>(_session)->shared_from_this());
 }
 
-/////////////////////////////////////////////////////////////////////////////////////////////
+void WebRtcTransportImp::setSession(Session *session) {
-
+    _session = session;
-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);
-}
-
-string WebRtcTransportImp::getIdentifier() const {
-    return StrPrinter << this;
 }

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
+    //udp session
-    Socket::Ptr _socket;
+    Session *_session;
     //推流的rtsp源
     RtspMediaSource::Ptr _push_src;
     unordered_map<string/*rid*/, RtspMediaSource::Ptr> _push_src_simulcast;