mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-10-31 08:39:34 +08:00
155 lines
5.4 KiB
C++
155 lines
5.4 KiB
C++
/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#include "WebRtcSession.h"
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#include "Util/util.h"
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#include "Network/TcpServer.h"
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#include "Common/config.h"
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#include "IceServer.hpp"
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#include "WebRtcTransport.h"
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using namespace std;
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namespace mediakit {
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static string getUserName(const char *buf, size_t len) {
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if (!RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
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return "";
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}
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std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *) buf, len));
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if (!packet) {
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return "";
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}
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if (packet->GetClass() != RTC::StunPacket::Class::REQUEST ||
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packet->GetMethod() != RTC::StunPacket::Method::BINDING) {
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return "";
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}
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//收到binding request请求
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auto vec = split(packet->GetUsername(), ":");
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return vec[0];
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}
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EventPoller::Ptr WebRtcSession::queryPoller(const Buffer::Ptr &buffer) {
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auto user_name = getUserName(buffer->data(), buffer->size());
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if (user_name.empty()) {
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return nullptr;
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}
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auto ret = WebRtcTransportManager::Instance().getItem(user_name);
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return ret ? ret->getPoller() : nullptr;
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}
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////////////////////////////////////////////////////////////////////////////////
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WebRtcSession::WebRtcSession(const Socket::Ptr &sock) : Session(sock) {
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_over_tcp = sock->sockType() == SockNum::Sock_TCP;
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}
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WebRtcSession::~WebRtcSession() = default;
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void WebRtcSession::attachServer(const Server &server) {
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_server = std::dynamic_pointer_cast<toolkit::TcpServer>(const_cast<Server &>(server).shared_from_this());
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}
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void WebRtcSession::onRecv_l(const char *data, size_t len) {
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if (_find_transport) {
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// 只允许寻找一次transport
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_find_transport = false;
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auto user_name = getUserName(data, len);
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auto transport = WebRtcTransportManager::Instance().getItem(user_name);
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CHECK(transport);
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//WebRtcTransport在其他poller线程上,需要切换poller线程并重新创建WebRtcSession对象
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if (!transport->getPoller()->isCurrentThread()) {
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auto sock = Socket::createSocket(transport->getPoller(), false);
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//1、克隆socket(fd不变),切换poller线程到WebRtcTransport所在线程
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sock->cloneSocket(*(getSock()));
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auto server = _server;
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std::string str(data, len);
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sock->getPoller()->async([sock, server, str](){
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auto strong_server = server.lock();
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if (strong_server) {
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auto session = static_pointer_cast<WebRtcSession>(strong_server->createSession(sock));
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//2、创建新的WebRtcSession对象(绑定到WebRtcTransport所在线程),重新处理一遍ice binding request命令
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session->onRecv_l(str.data(), str.size());
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}
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});
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//3、销毁原先的socket和WebRtcSession(原先的对象跟WebRtcTransport不在同一条线程)
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throw std::runtime_error("webrtc over tcp change poller: " + getPoller()->getThreadName() + " -> " + sock->getPoller()->getThreadName());
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}
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_transport = std::move(transport);
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InfoP(this);
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}
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_ticker.resetTime();
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CHECK(_transport);
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_transport->inputSockData((char *)data, len, this);
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}
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void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
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if (_over_tcp) {
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input(buffer->data(), buffer->size());
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} else {
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onRecv_l(buffer->data(), buffer->size());
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}
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}
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void WebRtcSession::onError(const SockException &err) {
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//udp链接超时,但是rtc链接不一定超时,因为可能存在链接迁移的情况
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//在udp链接迁移时,新的WebRtcSession对象将接管WebRtcTransport对象的生命周期
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//本WebRtcSession对象将在超时后自动销毁
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WarnP(this) << err;
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if (!_transport) {
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return;
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}
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auto self = static_pointer_cast<WebRtcSession>(shared_from_this());
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auto transport = std::move(_transport);
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getPoller()->async([transport, self]() mutable {
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//延时减引用,防止使用transport对象时,销毁对象
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transport->removeTuple(self.get());
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//确保transport在Session对象前销毁,防止WebRtcTransport::onDestory()时获取不到Session对象
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transport = nullptr;
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}, false);
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}
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void WebRtcSession::onManager() {
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GET_CONFIG(float, timeoutSec, Rtc::kTimeOutSec);
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if (!_transport && _ticker.createdTime() > timeoutSec * 1000) {
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shutdown(SockException(Err_timeout, "illegal webrtc connection"));
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return;
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}
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if (_ticker.elapsedTime() > timeoutSec * 1000) {
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shutdown(SockException(Err_timeout, "webrtc connection timeout"));
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return;
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}
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}
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ssize_t WebRtcSession::onRecvHeader(const char *data, size_t len) {
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onRecv_l(data + 2, len - 2);
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return 0;
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}
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const char *WebRtcSession::onSearchPacketTail(const char *data, size_t len) {
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if (len < 2) {
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// 数据不够
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return nullptr;
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}
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uint16_t length = (((uint8_t *)data)[0] << 8) | ((uint8_t *)data)[1];
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if (len < (size_t)(length + 2)) {
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// 数据不够
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return nullptr;
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}
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// 返回rtp包末尾
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return data + 2 + length;
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}
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}// namespace mediakit
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