Add UMA metrics and finch experiment for DTLS1.2 in WebRTC.

content/renderer/media/rtc_peer_connection_handler.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/lazy_instance.h"
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/metrics/histogram.h"
+#include "base/metrics/sparse_histogram.h"
 #include "base/stl_util.h"
 #include "base/strings/utf_string_conversions.h"
 #include "base/thread_task_runner_handle.h"
 #include "base/trace_event/trace_event.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
+#include "net/third_party/nss/ssl/sslproto.h"

[Ryan Sleevi, 2015/09/22 21:21:10]

BUG: You should not be directly including NSS headers, especially since Chrome
now uses BoringSSL.

 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/libjingle/source/talk/session/media/mediasession.h"
@@ skipping 11 matching lines @@
 
 // Used to back histogram value of "WebRTC.PeerConnection.RtcpMux",
 // so treat as append-only.
 enum RtcpMux {
   RTCP_MUX_DISABLED,
   RTCP_MUX_ENABLED,
   RTCP_MUX_NO_MEDIA,
   RTCP_MUX_MAX
 };
 
+#define SSLCIPHER_ENTRY(cipher) \
+  { webrtc::SslCipherType_##cipher, cipher }
+
+uint16 ConvertWebRtcSslCipherToStandard(int cipher) {
+  static struct {
+    webrtc::SslCipherType webrtc_cipher;
+    uint16 cipher;
+  } kWebRtcSslCipherMapping[] = {
+      // TLS v1.0 ciphersuites from RFC2246.
+      {webrtc::SslCipherType_TLS_RSA_RC4_128_SHA, TLS_RSA_WITH_RC4_128_SHA},
+      SSLCIPHER_ENTRY(TLS_RSA_WITH_3DES_EDE_CBC_SHA),
+
+      // AES ciphersuites from RFC3268.
+      SSLCIPHER_ENTRY(TLS_RSA_WITH_AES_128_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_DHE_RSA_WITH_AES_128_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_RSA_WITH_AES_256_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_DHE_RSA_WITH_AES_256_CBC_SHA),
+
+      // ECC ciphersuites from RFC4492.
+      SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_RC4_128_SHA),
+      SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_3DES_EDE_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA),
+
+      SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_RC4_128_SHA),
+      SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_3DES_EDE_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA),
+      SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA),
+
+      // TLS v1.2 ciphersuites.
+      SSLCIPHER_ENTRY(TLS_RSA_WITH_AES_128_CBC_SHA256),
+      SSLCIPHER_ENTRY(TLS_RSA_WITH_AES_256_CBC_SHA256),
+      SSLCIPHER_ENTRY(TLS_DHE_RSA_WITH_AES_128_CBC_SHA256),
+      SSLCIPHER_ENTRY(TLS_DHE_RSA_WITH_AES_256_CBC_SHA256),
+
+      // TLS v1.2 GCM ciphersuites from RFC5288.
+      SSLCIPHER_ENTRY(TLS_RSA_WITH_AES_128_GCM_SHA256),
+      // SSLCIPHER_ENTRY(TLS_RSA_WITH_AES_256_GCM_SHA384),
+      SSLCIPHER_ENTRY(TLS_DHE_RSA_WITH_AES_128_GCM_SHA256),
+      // SSLCIPHER_ENTRY(TLS_DHE_RSA_WITH_AES_256_GCM_SHA384),
+      // SSLCIPHER_ENTRY(TLS_DH_RSA_WITH_AES_128_GCM_SHA256),
+      // SSLCIPHER_ENTRY(TLS_DH_RSA_WITH_AES_256_GCM_SHA384),
+
+      // ECDH HMAC based ciphersuites from RFC5289.
+      SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA256),
+      // SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA384),
+      SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA256),
+      // SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384),
+
+      // ECDH GCM based ciphersuites from RFC5289.
+      SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256),
+      // SSLCIPHER_ENTRY(TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384),
+      SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256),
+      // SSLCIPHER_ENTRY(TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384),
+  };
+
+  for (auto c : kWebRtcSslCipherMapping) {
+    if (c.webrtc_cipher == cipher)
+      return c.cipher;
+  }

[Ryan Sleevi, 2015/09/22 21:21:10]

This is very inefficient for what it does - given that you've got a static
table, in source, you should be able to easily pre-sort this table and then use
either std::lower_bound or std::binary_search as appropriate.

Alternatively, if you simply define the WebRTC enum values to match their TLS
wire encoding, you avoid the need for any such mapping at all.

[davidben, 2015/09/22 21:29:46]

+1 to that. TLS already provides standard well-known values for each of these.
Please don't add a new enum. New ciphers may be added in the future and I don't
want to have to chase down multiple enums just to add them. (The WebRTC enum
cites kSslCipherMap which isn't even used in BoringSSL!)

+  return 0;
+}
+
 // Converter functions from libjingle types to WebKit types.
 blink::WebRTCPeerConnectionHandlerClient::ICEGatheringState
 GetWebKitIceGatheringState(
     webrtc::PeerConnectionInterface::IceGatheringState state) {
   using blink::WebRTCPeerConnectionHandlerClient;
   switch (state) {
     case webrtc::PeerConnectionInterface::kIceGatheringNew:
       return WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew;
     case webrtc::PeerConnectionInterface::kIceGatheringGathering:
       return WebRTCPeerConnectionHandlerClient::ICEGatheringStateGathering;
@@ skipping 449 matching lines @@
                                   counter_max);
         break;
       case webrtc::kEnumCounterIceCandidatePairTypeUdp:
         UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
                                   counter, counter_max);
         break;
       case webrtc::kEnumCounterIceCandidatePairTypeTcp:
         UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
                                   counter, counter_max);
         break;
+      case webrtc::kEnumCounterAudioSrtpCipher:
+        UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.Cipher.Audio.Srtp",
+                                  counter, counter_max);
+        break;
+      case webrtc::kEnumCounterAudioSslCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY("WebRTC.PeerConnection.Cipher.Audio.Ssl",
+                                    ConvertWebRtcSslCipherToStandard(counter));
+        break;
+      case webrtc::kEnumCounterVideoSrtpCipher:
+        UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.Cipher.Video.Srtp",
+                                  counter, counter_max);
+        break;
+      case webrtc::kEnumCounterVideoSslCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY("WebRTC.PeerConnection.Cipher.Video.Ssl",
+                                    ConvertWebRtcSslCipherToStandard(counter));
+        break;
+      case webrtc::kEnumCounterDataSrtpCipher:
+        UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.Cipher.Data.Srtp",
+                                  counter, counter_max);
+        break;
+      case webrtc::kEnumCounterDataSslCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY("WebRTC.PeerConnection.Cipher.Data.Ssl",
+                                    ConvertWebRtcSslCipherToStandard(counter));
+        break;
       default:
-        NOTREACHED();
         break;
     }
   }
 
   void AddHistogramSample(webrtc::PeerConnectionUMAMetricsName type,
                           int value) override {
     // Runs on libjingle's signaling thread.
     switch (type) {
       case webrtc::kTimeToConnect:
         UMA_HISTOGRAM_MEDIUM_TIMES(
@@ skipping 1065 matching lines @@
 }
 
 void RTCPeerConnectionHandler::ResetUMAStats() {
   DCHECK(thread_checker_.CalledOnValidThread());
   num_local_candidates_ipv6_ = 0;
   num_local_candidates_ipv4_ = 0;
   ice_connection_checking_start_ = base::TimeTicks();
   memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
 }
 }  // namespace content