Change WebRTC SslCipher to be exposed as number only.

webrtc/p2p/base/p2ptransportchannel.h

 /*
  *  Copyright 2004 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 93 matching lines @@
   bool GetSslRole(rtc::SSLRole* role) const override { return false; }
 
   bool SetSslRole(rtc::SSLRole role) override { return false; }
 
   // Set up the ciphers to use for DTLS-SRTP.
   bool SetSrtpCiphers(const std::vector<std::string>& ciphers) override {
     return false;
   }
 
   // Find out which DTLS-SRTP cipher was negotiated.
-  bool GetSrtpCipher(std::string* cipher) override { return false; }
+  bool GetSrtpCryptoSuite(std::string* cipher) override { return false; }
 
   // Find out which DTLS cipher was negotiated.
-  bool GetSslCipher(std::string* cipher) override { return false; }
+  bool GetSslCipherSuite(uint16_t* cipher) override { return false; }
 
   // Returns null because the channel is not encrypted by default.
   rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override {
     return nullptr;
   }
 
   bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert) const override {
     return false;
   }
 
@@ skipping 123 matching lines @@
   int receiving_timeout_;
   uint32 last_ping_sent_ms_ = 0;
   bool gather_continually_ = false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_