/* * 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" static string getUserName(const Buffer::Ptr &buffer) { auto buf = buffer->data(); auto len = buffer->size(); if (!RTC::StunPacket::IsStun((const uint8_t *) buf, len)) { return ""; } std::unique_ptr packet(RTC::StunPacket::Parse((const uint8_t *) buf, len)); if (!packet) { return ""; } if (packet->GetClass() != RTC::StunPacket::Class::REQUEST || packet->GetMethod() != RTC::StunPacket::Method::BINDING) { return ""; } //收到binding request请求 auto vec = split(packet->GetUsername(), ":"); return vec[0]; } EventPoller::Ptr WebRtcSession::queryPoller(const Buffer::Ptr &buffer) { auto user_name = getUserName(buffer); if (user_name.empty()) { return nullptr; } auto ret = WebRtcTransportManager::Instance().getItem(user_name); return ret ? ret->getPoller() : nullptr; } //////////////////////////////////////////////////////////////////////////////// WebRtcSession::WebRtcSession(const Socket::Ptr &sock) : UdpSession(sock) { socklen_t addr_len = sizeof(_peer_addr); getpeername(sock->rawFD(), &_peer_addr, &addr_len); } WebRtcSession::~WebRtcSession() { InfoP(this); } void WebRtcSession::onRecv(const Buffer::Ptr &buffer) { if (_find_transport) { //只允许寻找一次transport _find_transport = false; auto user_name = getUserName(buffer); _identifier = to_string(getSock()->rawFD()) + '-' + user_name; auto transport = WebRtcTransportManager::Instance().getItem(user_name); CHECK(transport && transport->getPoller()->isCurrentThread()); transport->setSession(shared_from_this()); _transport = std::move(transport); InfoP(this); } _ticker.resetTime(); CHECK(_transport); _transport->inputSockData(buffer->data(), buffer->size(), &_peer_addr); } void WebRtcSession::onError(const SockException &err) { //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 //在udp链接迁移时,新的WebRtcSession对象将接管WebRtcTransport对象的生命周期 //本WebRtcSession对象将在超时后自动销毁 WarnP(this) << err.what(); if (!_transport) { return; } auto transport = std::move(_transport); getPoller()->async([transport] { //延时减引用,防止使用transport对象时,销毁对象 }, false); } void WebRtcSession::onManager() { GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec); if (!_transport && _ticker.createdTime() > timeoutSec * 1000) { shutdown(SockException(Err_timeout, "illegal webrtc connection")); return; } if (_ticker.elapsedTime() > timeoutSec * 1000) { shutdown(SockException(Err_timeout, "webrtc connection timeout")); return; } } std::string WebRtcSession::getIdentifier() const { return _identifier; }