mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
106 lines
3.3 KiB
C++
106 lines
3.3 KiB
C++
/*
|
||
* 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);
|
||
}
|
||
|
||
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 QueryPollerByBuffer(const Buffer::Ptr &buffer) {
|
||
auto user_name = getUserName(buffer);
|
||
if (user_name.empty()) {
|
||
return nullptr;
|
||
}
|
||
auto ret = WebRtcTransportImp::get(user_name);
|
||
return ret ? ret->getPoller() : nullptr;
|
||
}
|
||
|
||
void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
|
||
try {
|
||
onRecv_l(buffer);
|
||
} catch (std::exception &ex) {
|
||
shutdown(SockException(Err_shutdown, ex.what()));
|
||
}
|
||
}
|
||
|
||
void WebRtcSession::onRecv_l(const Buffer::Ptr &buffer) {
|
||
if (_find_transport) {
|
||
//只允许寻找一次transport
|
||
_find_transport = false;
|
||
auto user_name = getUserName(buffer);
|
||
_identifier = user_name + '-' + to_string(reinterpret_cast<uint64_t>(this));
|
||
_transport = WebRtcTransportImp::move(user_name);
|
||
CHECK(_transport && _transport->getPoller()->isCurrentThread());
|
||
_transport->setSession(shared_from_this());
|
||
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;
|
||
}
|
||
|