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webrtc单端口模式支持线程安全，支持链接迁移

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This commit is contained in:
ziyue 2021-09-10 18:37:32 +08:00
parent c025b5092d
commit 7ba44d1ac8
4 changed files with 55 additions and 27 deletions
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2
3rdpart/ZLToolKit

@ -1 +1 @@
Subproject commit a827f0bdcb4221c19ab1a243ebd1f01fa7dc50f2
Subproject commit b8e066222b757a2c11b5e44c49ef6982acb95fe2

11
server/main.cpp
View File

@ -309,6 +309,17 @@ int start_main(int argc,char *argv[]) {
#if defined(ENABLE_WEBRTC)
//webrtc udp服务器
UdpServer::Ptr rtcSrv = std::make_shared<UdpServer>();
rtcSrv->setOnCreateSocket([](const EventPoller::Ptr &poller, const Buffer::Ptr &buf, struct sockaddr *, int) {
if (!buf) {
return Socket::createSocket(poller, false);
}
auto new_poller = WebRtcSession::getPoller(buf);
if (!new_poller) {
//该数据对应的webrtc对象未找到，丢弃之
return Socket::Ptr();
}
return Socket::createSocket(new_poller, false);
});
uint16_t rtcPort = mINI::Instance()[RTC::kPort];
#endif//defined(ENABLE_WEBRTC)

62
webrtc/WebRtcSession.cpp
View File

@ -21,26 +21,54 @@ 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;
}
auto ret = WebRtcTransportImp::getTransport(user_name);
return ret ? ret->getPoller() : nullptr;
}
void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
auto buf = buffer->data();
auto len = buffer->size();
if (!_transport && RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *) buf, len));
if (!packet) {
WarnL << "parse stun error";
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;
}
if (packet->GetClass() == RTC::StunPacket::Class::REQUEST &&
packet->GetMethod() == RTC::StunPacket::Method::BINDING) {
//收到binding request请求
_transport = createTransport(packet->GetUsername());
_transport = WebRtcTransportImp::getTransport(user_name);
if (!_transport) {
//逻辑分支不太可能走到这里
WarnL << user_name;
return;
}
_transport->setSession(this);
}
if (_transport) {
_transport->inputSockData(buf, len, &_peer_addr);
}
_transport->inputSockData(buf, len, &_peer_addr);
}
void WebRtcSession::onError(const SockException &err) {
@ -53,13 +81,3 @@ void WebRtcSession::onError(const SockException &err) {
void WebRtcSession::onManager() {
}
std::shared_ptr<WebRtcTransport> WebRtcSession::createTransport(const string &user_name) {
if (user_name.empty()) {
return nullptr;
}
auto vec = split(user_name, ":");
auto ret = WebRtcTransportImp::getTransport(vec[0]);
ret->setSession(this);
return ret;
}

7
webrtc/WebRtcSession.h
View File

@ -20,6 +20,8 @@ using namespace toolkit;
class WebRtcSession : public UdpSession {
public:
static EventPoller::Ptr getPoller(const Buffer::Ptr &);
WebRtcSession(const Socket::Ptr &sock);
~WebRtcSession() override;
@ -27,12 +29,9 @@ public:
void onError(const SockException &err) override;
void onManager() override;
private:
std::shared_ptr<WebRtcTransport> createTransport(const string &user_name);
private:
struct sockaddr _peer_addr;
std::shared_ptr<WebRtcTransport> _transport;
std::shared_ptr<WebRtcTransportImp> _transport;
};