Implement RTP keepalive in native stack.

webrtc/video_send_stream.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 150 matching lines @@
       // details.
       struct Rtx {
         std::string ToString() const;
         // SSRCs to use for the RTX streams.
         std::vector<uint32_t> ssrcs;
 
         // Payload type to use for the RTX stream.
         int payload_type = -1;
       } rtx;
 
+      struct KeepAlive {
+        // If no packet has been sent for |timeout_interval_ms|, send a keep-
+        // alive packet. The keep-laive packet is an empty (no payload) RTP

[åsapersson, 2017/06/30 15:09:13]

laive->alive

+        // packet with a payload type of 20 as long as the other end has not
+        // negotiated the use of this value. If this value has already been
+        // negotiated, then some other unused static payload type from table 5
+        // of RFC 3551 shall be used and set in |payload_type|.
+        int64_t timeout_interval_ms = -1;
+        uint8_t payload_type = 20;
+      } keep_alive;
+
       // RTCP CNAME, see RFC 3550.
       std::string c_name;
     } rtp;
 
     // Transport for outgoing packets.
     Transport* send_transport = nullptr;
 
     // Called for each I420 frame before encoding the frame. Can be used for
     // effects, snapshots etc. 'nullptr' disables the callback.
     rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
@@ skipping 78 matching lines @@
     EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
   }
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_