Move RTP keep-alive config from VideoSendStream::Config to Call::Config

webrtc/call/call.h

Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index 86142f0feaaf065c3215a9e3086bfdf63b08911e..f99ba851ab420daeb5c12f03e1577f1701c4cbb3 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -112,6 +112,12 @@ class Call {
     // RtcEventLog to use for this call. Required.
     // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
     RtcEventLog* event_log = nullptr;
+
+    // Enables periodic sending if empty keep-alive messages that helps prevent
+    // network time-out events. The packets adhere to RFC6263 section 4.6, and
+    // by default use payload type 20, as described in 3GPP TS 24.229,
+    // Appendix K.5.2.1.
+    RtpKeepAliveConfig keepalive_config;
   };
 
   struct Stats {