RtpRtcp allows to register header extension by name

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
 #include <list>
 #include <set>
+#include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
@@ skipping 26 matching lines @@
   int32_t RegisterSendPayload(const VideoCodec& video_codec) override;
 
   int32_t DeRegisterSendPayload(int8_t payload_type) override;
 
   int8_t SendPayloadType() const;
 
   // Register RTP header extension.
   int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
                                          uint8_t id) override;
 
+  bool RegisterRtpHeaderExtension(const std::string& type, uint8_t id) override;
+
   int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
 
   // Get start timestamp.
   uint32_t StartTimestamp() const override;
 
   // Configure start timestamp, default is a random number.
   void SetStartTimestamp(uint32_t timestamp) override;
 
   uint16_t SequenceNumber() const override;
 
@@ skipping 308 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_