Fix spelling mistake in rtp_rtcp.h.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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.
  */
 
@@ skipping 100 matching lines @@
 
   // **************************************************************************
   // Sender
   // **************************************************************************
 
   // Sets MTU.
   // |size| - Max transfer unit in bytes, default is 1500.
   // Returns -1 on failure else 0.
   virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
 
-  // Sets transtport overhead. Default is IPv4 and UDP with no encryption.
+  // Sets transport overhead. Default is IPv4 and UDP with no encryption.
   // |tcp| - true for TCP false UDP.
   // |ipv6| - true for IP version 6 false for version 4.
   // |authentication_overhead| - number of bytes to leave for an authentication
   // header.
   // Returns -1 on failure else 0
   // TODO(michaelt): deprecate the function.
   virtual int32_t SetTransportOverhead(bool tcp,
                                        bool ipv6,
                                        uint8_t authentication_overhead = 0) = 0;
 
@@ skipping 345 matching lines @@
   virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
 
   // Sends a request for a keyframe.
   // Returns -1 on failure else 0.
   virtual int32_t RequestKeyFrame() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_