ZLMediaKit/webrtc/WebRtcSession.cpp

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/*
* 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<RTC::StunPacket> 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;
}
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auto ret = WebRtcTransportImp::getRtcTransport(user_name, false);
return ret ? ret->getPoller() : nullptr;
}
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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;
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return;
}
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_transport = WebRtcTransportImp::getRtcTransport(user_name, true);
if (!_transport) {
//逻辑分支不太可能走到这里
WarnL << user_name;
return;
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}
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_transport->setSession(shared_from_this());
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}
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_ticker.resetTime();
_transport->inputSockData(buf, len, &_peer_addr);
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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
//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;
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}
void WebRtcSession::onManager() {
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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;
}
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}