Index: webrtc/config.cc |
diff --git a/webrtc/config.cc b/webrtc/config.cc |
index ddff931e241fc1ea22f01b09dd149c754eb9686c..3a74e5238e7d831a145ae97d33d54f6d76b10c51 100644 |
--- a/webrtc/config.cc |
+++ b/webrtc/config.cc |
@@ -36,7 +36,7 @@ const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation"; |
const char* RtpExtension::kAudioLevel = |
"urn:ietf:params:rtp-hdrext:ssrc-audio-level"; |
const char* RtpExtension::kTransportSequenceNumber = |
- "http://www.webrtc.org/experiments/rtp-hdrext/transport-sequence-number"; |
+ "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions"; |
bool RtpExtension::IsSupportedForAudio(const std::string& name) { |
return name == webrtc::RtpExtension::kAbsSendTime || |