接受和发送的数据处理tcp 2字节头

webrtc/WebRtcSession.cpp

@@ -16,8 +16,8 @@ using namespace std;
 namespace mediakit {
 
 static string getUserName(const Buffer::Ptr &buffer) {
-    auto buf = buffer->data();
-    auto len = buffer->size();
+    auto buf = buffer->data() + 2;
+    auto len = buffer->size() - 2;
     if (!RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
         return "";
     }
@@ -55,9 +55,8 @@ WebRtcSession::~WebRtcSession() {
 }
 
 void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
-    if (_find_transport) {
-        //只允许寻找一次transport
-        _find_transport = false;
+    //只允许寻找一次transport
+    if (!_transport) {
         auto user_name = getUserName(buffer);
         auto transport = WebRtcTransportManager::Instance().getItem(user_name);
         //TODO fix this poller is not current thread
@@ -68,7 +67,7 @@ void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
     }
     _ticker.resetTime();
     CHECK(_transport);
-    _transport->inputSockData(buffer->data(), buffer->size(), (struct sockaddr *)&_peer_addr);
+    _transport->inputSockData(buffer->data() + 2, buffer->size() - 2, (struct sockaddr *)&_peer_addr);
 }
 
 void WebRtcSession::onError(const SockException &err) {

webrtc/WebRtcSession.h

@@ -31,7 +31,6 @@ public:
     static EventPoller::Ptr queryPoller(const Buffer::Ptr &buffer);
 
 private:
-    bool _find_transport = true;
     Ticker _ticker;
     struct sockaddr_storage _peer_addr;
     std::shared_ptr<WebRtcTransportImp> _transport;

webrtc/WebRtcTransport.cpp

@@ -418,6 +418,18 @@ void WebRtcTransportImp::onSendSockData(Buffer::Ptr buf, bool flush, RTC::Transp
         WarnL << "send data failed:" << buf->size();
         return;
     }
+
+    //增加tcp两字节头
+    auto pkt = _packet_pool.obtain2();
+    int len = buf->size();
+    pkt->setCapacity(buf->size() + 2 + 1);
+    char tcp_len[2] = {0};
+    tcp_len[0] = ((int16_t)len >> 8)&0xff;
+    tcp_len[1] = (int16_t)len & 0xff;
+    pkt->assign2(tcp_len, 2);
+    pkt->assign2(buf->data(), len , 2);
+    pkt->setSize(len + 2);
+
     // 一次性发送一帧的rtp数据，提高网络io性能
     _selected_session->setSendFlushFlag(flush);
     _selected_session->send(std::move(buf));

webrtc/WebRtcTransport.h

@@ -127,6 +127,9 @@ protected:
     void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
     void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
 
+    // 循环池
+    ResourcePool<BufferRaw> _packet_pool;
+
 protected:
     //// ice相关的回调 ///
     void OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) override;
@@ -179,8 +182,6 @@ private:
     std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
     std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
     Ticker _ticker;
-    //循环池
-    ResourcePool<BufferRaw> _packet_pool;
 
 #ifdef ENABLE_SCTP
     RTC::SctpAssociationImp::Ptr _sctp;