Implement FrameMarking header extension support

webrtc/config.cc

Index: webrtc/config.cc
diff --git a/webrtc/config.cc b/webrtc/config.cc
index 36e9c3ab9a92e62a97e658087e0d973a47ab38dc..ff7b0cc47dc49304fb6b53eea08667d32025446c 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -84,6 +84,13 @@ const char* RtpExtension::kVideoTimingUri =
     "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
 const int RtpExtension::kVideoTimingDefaultId = 8;
 
+// This extensions provides meta-information about the RTP streams outside the
+// encrypted media payload, an RTP switch can do codec-agnostic
+// selective forwarding without decrypting the payload
+const char* RtpExtension::kFrameMarkingUri =
+    "urn:ietf:params:rtp-hdrext:framemarking";
+const int RtpExtension::kFrameMarkingDefaultId = 9;
+
 const char* RtpExtension::kEncryptHeaderExtensionsUri =
     "urn:ietf:params:rtp-hdrext:encrypt";
 
@@ -102,7 +109,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
-         uri == webrtc::RtpExtension::kVideoTimingUri;
+         uri == webrtc::RtpExtension::kVideoTimingUri ||
+         uri == webrtc::RtpExtension::kFrameMarkingUri;
 }
 
 bool RtpExtension::IsEncryptionSupported(const std::string& uri) {