mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-23 11:17:09 +08:00
c72cf4cbcc
* feat: remove using namespace mediakit in header files. (cherry picked from commit d44aeb339a8a0e1f0455be82b21fe4b1b536299f) * feat: remove using namespace mediakit in FFmpegSource.h * feat: remove using namespace mediakit in RtpExt.h * feat: remove using namespace mediakit in header files. * feat: remove using namespace std in header files. * feat: remove using namespace std in header files when zltoolkit remove std in header * 补充命名空间 * 整理命名空间 * 整理命名空间2 * 修复macos ci * 修复编译问题 * 修复编译问题2 * 修复编译问题3 Co-authored-by: Johnny <hellojinqiang@gmail.com> Co-authored-by: Xiaofeng Wang <wasphin@gmail.com>
104 lines
3.3 KiB
C++
104 lines
3.3 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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using namespace std;
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using namespace mediakit;
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static string getUserName(const Buffer::Ptr &buffer) {
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auto buf = buffer->data();
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auto len = buffer->size();
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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);
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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) : UdpSession(sock) {
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socklen_t addr_len = sizeof(_peer_addr);
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getpeername(sock->rawFD(), &_peer_addr, &addr_len);
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}
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WebRtcSession::~WebRtcSession() {
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InfoP(this);
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}
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void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
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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(buffer);
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_identifier = to_string(getSock()->rawFD()) + '-' + user_name;
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auto transport = WebRtcTransportManager::Instance().getItem(user_name);
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CHECK(transport && transport->getPoller()->isCurrentThread());
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transport->setSession(shared_from_this());
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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(buffer->data(), buffer->size(), &_peer_addr);
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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udp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.what();
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if (!_transport) {
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return;
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}
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auto transport = std::move(_transport);
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getPoller()->async([transport] {
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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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}, 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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std::string WebRtcSession::getIdentifier() const {
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return _identifier;
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}
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