初步实现webrtc单udp端口模式

webrtc/WebRtcSession.cpp

@@ -0,0 +1,65 @@
+/*
+ * 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);
+}
+
+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";
+            return;
+        }
+        if (packet->GetClass() == RTC::StunPacket::Class::REQUEST &&
+            packet->GetMethod() == RTC::StunPacket::Method::BINDING) {
+            //收到binding request请求
+            _transport = createTransport(packet->GetUsername());
+        }
+    }
+
+    if (_transport) {
+        _transport->inputSockData(buf, len, &_peer_addr);
+    }
+}
+
+void WebRtcSession::onError(const SockException &err) {
+    if (_transport) {
+        _transport->unrefSelf(err);
+        _transport = nullptr;
+    }
+}
+
+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;
+}