/* * 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); } 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::getPoller(const Buffer::Ptr &buffer) { auto user_name = getUserName(buffer); if (user_name.empty()) { return nullptr; } auto ret = WebRtcTransportImp::getRtcTransport(user_name, false); return ret ? ret->getPoller() : nullptr; } void WebRtcSession::onRecv(const Buffer::Ptr &buffer) { auto buf = buffer->data(); auto len = buffer->size(); if (!_transport) { auto user_name = getUserName(buffer); if (user_name.empty()) { //逻辑分支不太可能走到这里 WarnL << user_name; return; } _transport = WebRtcTransportImp::getRtcTransport(user_name, true); if (!_transport) { //逻辑分支不太可能走到这里 WarnL << user_name; return; } _transport->setSession(shared_from_this()); } _ticker.resetTime(); _transport->inputSockData(buf, len, &_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(); _transport = nullptr; } 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; } }