Implement RTP keepalive in native stack.

webrtc/video_send_stream.h

Index: webrtc/video_send_stream.h
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 266112e5871550d9406a945981647724170d1bfa..783a16bcda30dbaf0f9d3a0ef6c237b33c104c7a 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -168,6 +168,17 @@ class VideoSendStream {
         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
+        // 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;