Removes rtp level keep alive support.

This is not used in practice as there's functionality on
other levels that serves the same purpose.

Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
diff --git a/api/ortc/rtp_transport_interface.h b/api/ortc/rtp_transport_interface.h
index ec71216..ac43831 100644
--- a/api/ortc/rtp_transport_interface.h
+++ b/api/ortc/rtp_transport_interface.h
@@ -26,12 +26,8 @@
 struct RtpTransportParameters final {
   RtcpParameters rtcp;
 
-  // Enabled periodic sending of keep-alive packets, that help prevent timeouts
-  // on the network level, such as NAT bindings. See RFC6263 section 4.6.
-  RtpKeepAliveConfig keepalive;
-
   bool operator==(const RtpTransportParameters& o) const {
-    return rtcp == o.rtcp && keepalive == o.keepalive;
+    return rtcp == o.rtcp;
   }
   bool operator!=(const RtpTransportParameters& o) const {
     return !(*this == o);
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
index 8ab560c..04c3853 100644
--- a/api/rtp_headers.h
+++ b/api/rtp_headers.h
@@ -171,23 +171,6 @@
   kNetworkDown,
 };
 
-struct RtpKeepAliveConfig final {
-  // If no packet has been sent for |timeout_interval_ms|, send a keep-alive
-  // packet. The keep-alive 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;
-
-  bool operator==(const RtpKeepAliveConfig& o) const {
-    return timeout_interval_ms == o.timeout_interval_ms &&
-           payload_type == o.payload_type;
-  }
-  bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
-};
-
 }  // namespace webrtc
 
 #endif  // API_RTP_HEADERS_H_