Wire up RTP keep-alive in ortc api.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 3966d5e0309add865df305fbf284f689c317400e..809e64617a8ef765f7bd47d6ba8d6bd11ec094d6 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -211,6 +211,7 @@ class Call : public webrtc::Call,
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
+  bool SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) override;
 
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps,
@@ -1141,6 +1142,20 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   transport_send_->send_side_cc()->OnSentPacket(sent_packet);
 }
 
+bool Call::SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) {
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+
+  ReadLockScoped lock(*send_crit_);
+  if (config != config_.keepalive_config &&
+      (!video_send_streams_.empty() || !audio_send_ssrcs_.empty())) {
+    LOG(LS_WARNING) << "RTP keep-alive settings cannot be altered after "
+                       "creating send streams.";
+    return false;
+  }
+  config_.keepalive_config = config;
+  return true;
+}
+
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt_ms,