拷贝后再修改rtp，防止修改共享数据

webrtc/SrtpSession.cpp

@@ -231,23 +231,11 @@ namespace RTC
 		}
 	}
 
-	bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt)
+	bool SrtpSession::EncryptRtp(uint8_t* data, size_t* len)
 	{
 		MS_TRACE();
-
-		// Ensure that the resulting SRTP packet fits into the encrypt buffer.
-		if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize)
-		{
-			MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%zu bytes)", *len);
-
-			return false;
-		}
-
-		std::memcpy(EncryptBuffer, *data, *len);
-        EncryptBuffer[1] = (pt & 0x7F) | (EncryptBuffer[1] & 0x80);
-
 		srtp_err_status_t err =
-		  srtp_protect(this->session, static_cast<void*>(EncryptBuffer), reinterpret_cast<int*>(len));
+		  srtp_protect(this->session, static_cast<void*>(data), reinterpret_cast<int*>(len));
 
 		if (DepLibSRTP::IsError(err))
 		{
@@ -256,9 +244,6 @@ namespace RTC
 			return false;
 		}
 
-		// Update the given data pointer.
-		*data = (const uint8_t*)EncryptBuffer;
-
 		return true;
 	}
 
@@ -279,22 +264,11 @@ namespace RTC
 		return true;
 	}
 
-	bool SrtpSession::EncryptRtcp(const uint8_t** data, size_t* len)
+	bool SrtpSession::EncryptRtcp(uint8_t* data, size_t* len)
 	{
 		MS_TRACE();
-
-		// Ensure that the resulting SRTCP packet fits into the encrypt buffer.
-		if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize)
-		{
-			MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%zu bytes)", *len);
-
-			return false;
-		}
-
-		std::memcpy(EncryptBuffer, *data, *len);
-
 		srtp_err_status_t err = srtp_protect_rtcp(
-		  this->session, static_cast<void*>(EncryptBuffer), reinterpret_cast<int*>(len));
+		  this->session, static_cast<void*>(data), reinterpret_cast<int*>(len));
 
 		if (DepLibSRTP::IsError(err))
 		{
@@ -303,9 +277,6 @@ namespace RTC
 			return false;
 		}
 
-		// Update the given data pointer.
-		*data = (const uint8_t*)EncryptBuffer;
-
 		return true;
 	}
 

webrtc/SrtpSession.hpp

@@ -64,9 +64,9 @@ namespace RTC
 		~SrtpSession();
 
 	public:
-		bool EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt);
+		bool EncryptRtp(uint8_t* data, size_t* len);
 		bool DecryptSrtp(uint8_t* data, size_t* len);
-		bool EncryptRtcp(const uint8_t** data, size_t* len);
+		bool EncryptRtcp(uint8_t* data, size_t* len);
 		bool DecryptSrtcp(uint8_t* data, size_t* len);
 		void RemoveStream(uint32_t ssrc)
 		{
@@ -76,9 +76,6 @@ namespace RTC
 	private:
 		// Allocated by this.
 		srtp_t session{ nullptr };
-		//rtp包最大1600
-        static constexpr size_t EncryptBufferSize{ 2000 };
-        uint8_t EncryptBuffer[EncryptBufferSize];
         DepLibSRTP::Ptr _env;
 	};
 } // namespace RTC

webrtc/WebRtcTransport.cpp

@@ -266,25 +266,24 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
     }
 }
 
-void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) {
-    const uint8_t *p = (uint8_t *) buf;
-    bool ret = false;
+void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, TrackType type) {
     if (_srtp_session_send) {
-        ret = _srtp_session_send->EncryptRtp(&p, &len, pt);
-    }
-    if (ret) {
-        onSendSockData((char *) p, len, flush);
+        CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
+        memcpy(_srtp_buf, buf, len);
+        onBeforeEncryptRtp((char *) _srtp_buf, len, type);
+        if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
+            onSendSockData((char *) _srtp_buf, len, flush);
+        }
     }
 }
 
 void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){
-    const uint8_t *p = (uint8_t *) buf;
-    bool ret = false;
     if (_srtp_session_send) {
-        ret = _srtp_session_send->EncryptRtcp(&p, &len);
-    }
-    if (ret) {
-        onSendSockData((char *) p, len, flush);
+        CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
+        memcpy(_srtp_buf, buf, len);
+        if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
+            onSendSockData((char *) _srtp_buf, len, flush);
+        }
     }
 }
 
@@ -611,6 +610,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
             }
             case RtcpType::RTCP_PSFB:
             case RtcpType::RTCP_RTPFB: {
+                DebugL << "\r\n" << rtcp->dumpString();
                 break;
             }
             default: break;
@@ -663,8 +663,7 @@ static void setExtType(RtpExt &ext, RtpExtType tp) {
 }
 
 template<typename Type>
-static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) {
-    auto header = rtp->getHeader();
+static void changeRtpExtId(const RtpHeader *header, const Type &map) {
     auto ext_map = RtpExt::getExtValue(header);
     for (auto &pr : ext_map) {
         auto it = map.find((typename Type::key_type) (pr.first));
@@ -675,29 +674,35 @@ static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) {
         }
         setExtType(pr.second, it->first);
         setExtType(pr.second, it->second);
+        DebugL << pr.second.dumpString();
         pr.second.setExtId((uint8_t) it->second);
     }
 }
 
 void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
-    changeRtpExtId(rtp, _rtp_ext_id_to_type);
+    changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type);
     //统计rtp收到的情况，好做rr汇报
     info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 }
 
 void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
-    auto &pt = _send_rtp_pt[rtp->type];
+    auto pt = _send_rtp_pt[rtp->type];
     if (pt == 0xFF) {
         //忽略，对方不支持该编码类型
         return;
     }
-    changeRtpExtId(rtp, _rtp_ext_type_to_id);
     _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
-    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
+    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type);
     //统计rtp发送情况，好做sr汇报
     _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 }
 
+void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) {
+    auto header = (RtpHeader *)buf;
+    header->pt = _send_rtp_pt[type];
+    changeRtpExtId(header, _rtp_ext_type_to_id);
+}
+
 void WebRtcTransportImp::onShutdown(const SockException &ex){
     InfoL << ex.what();
     _self = nullptr;

webrtc/WebRtcTransport.h

@@ -60,9 +60,9 @@ public:
      * @param buf rtcp内容
      * @param len rtcp长度
      * @param flush 是否flush socket
-     * @param pt rtp payload type
+     * @param type rtp类型
      */
-    void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt);
+    void sendRtpPacket(char *buf, size_t len, bool flush, TrackType type);
     void sendRtcpPacket(char *buf, size_t len, bool flush);
 
     const EventPoller::Ptr& getPoller() const;
@@ -100,6 +100,7 @@ protected:
     virtual void onRtp(const char *buf, size_t len) = 0;
     virtual void onRtcp(const char *buf, size_t len) = 0;
     virtual void onShutdown(const SockException &ex) = 0;
+    virtual void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) = 0;
 
 protected:
     const RtcSession& getSdp(SdpType type) const;
@@ -112,6 +113,7 @@ private:
     void setRemoteDtlsFingerprint(const RtcSession &remote);
 
 private:
+    uint8_t _srtp_buf[2000];
     EventPoller::Ptr _poller;
     std::shared_ptr<RTC::IceServer> _ice_server;
     std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
@@ -150,6 +152,8 @@ protected:
 
     void onRtp(const char *buf, size_t len) override;
     void onRtcp(const char *buf, size_t len) override;
+    void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) override;
+
     void onShutdown(const SockException &ex) override;
 
     ///////MediaSourceEvent override///////