Expose negotiated ciphers through stats API.
Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.
BUG=3976
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35169004
Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index d6b9bf2..8e1c43f 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -102,6 +102,15 @@
static const char kAudioTrackLabelBase[] = "audio_track";
static const char kDataChannelLabel[] = "data_channel";
+// Disable for TSan v2, see
+// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
+// This declaration is also #ifdef'd as it causes unused-variable errors.
+#if !defined(THREAD_SANITIZER)
+// SRTP cipher name negotiated by the tests. This must be updated if the
+// default changes.
+static const char kDefaultSrtpCipher[] = "AES_CM_128_HMAC_SHA1_32";
+#endif
+
static void RemoveLinesFromSdp(const std::string& line_start,
std::string* sdp) {
const char kSdpLineEnd[] = "\r\n";
@@ -380,6 +389,24 @@
return bw;
}
+ std::string GetDtlsCipherStats() {
+ rtc::scoped_refptr<MockStatsObserver>
+ observer(new rtc::RefCountedObject<MockStatsObserver>());
+ EXPECT_TRUE(peer_connection_->GetStats(
+ observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
+ EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+ return observer->DtlsCipher();
+ }
+
+ std::string GetSrtpCipherStats() {
+ rtc::scoped_refptr<MockStatsObserver>
+ observer(new rtc::RefCountedObject<MockStatsObserver>());
+ EXPECT_TRUE(peer_connection_->GetStats(
+ observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
+ EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+ return observer->SrtpCipher();
+ }
+
int rendered_width() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty() ? 1 :
@@ -1280,6 +1307,22 @@
kMaxWaitForStatsMs);
}
+// Test that we can get negotiated ciphers.
+TEST_F(JsepPeerConnectionP2PTestClient, GetNegotiatedCiphersStats) {
+ ASSERT_TRUE(CreateTestClients());
+ LocalP2PTest();
+
+ EXPECT_EQ_WAIT(
+ rtc::SSLStreamAdapter::GetDefaultSslCipher(),
+ initializing_client()->GetDtlsCipherStats(),
+ kMaxWaitForStatsMs);
+
+ EXPECT_EQ_WAIT(
+ kDefaultSrtpCipher,
+ initializing_client()->GetSrtpCipherStats(),
+ kMaxWaitForStatsMs);
+}
+
// This test sets up a call between two parties with audio, video and data.
TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
FakeConstraints setup_constraints;
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index 968fa8f..d434aaf 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -715,6 +715,18 @@
StatsReport::kStatsValueNameRemoteCertificateId,
remote_cert_report_id);
}
+ const std::string& srtp_cipher = channel_iter->srtp_cipher;
+ if (!srtp_cipher.empty()) {
+ channel_report->AddValue(
+ StatsReport::kStatsValueNameSrtpCipher,
+ srtp_cipher);
+ }
+ const std::string& ssl_cipher = channel_iter->ssl_cipher;
+ if (!ssl_cipher.empty()) {
+ channel_report->AddValue(
+ StatsReport::kStatsValueNameDtlsCipher,
+ ssl_cipher);
+ }
for (size_t i = 0;
i < channel_iter->connection_infos.size();
++i) {
@@ -742,8 +754,6 @@
info.writable);
report->AddBoolean(StatsReport::kStatsValueNameReadable,
info.readable);
- report->AddBoolean(StatsReport::kStatsValueNameActiveConnection,
- info.best_connection);
report->AddValue(StatsReport::kStatsValueNameLocalCandidateId,
AddCandidateReport(info.local_candidate, true));
report->AddValue(
@@ -760,6 +770,13 @@
info.local_candidate.type());
report->AddValue(StatsReport::kStatsValueNameRemoteCandidateType,
info.remote_candidate.type());
+ report->AddBoolean(StatsReport::kStatsValueNameActiveConnection,
+ info.best_connection);
+ if (info.best_connection) {
+ channel_report->AddValue(
+ StatsReport::kStatsValueNameSelectedCandidatePairId,
+ report->id().ToString());
+ }
}
}
}
diff --git a/talk/app/webrtc/statscollector_unittest.cc b/talk/app/webrtc/statscollector_unittest.cc
index 8a1dc77..8fd59e9 100644
--- a/talk/app/webrtc/statscollector_unittest.cc
+++ b/talk/app/webrtc/statscollector_unittest.cc
@@ -644,6 +644,8 @@
// Fake stats to process.
cricket::TransportChannelStats channel_stats;
channel_stats.component = 1;
+ channel_stats.srtp_cipher = "the-srtp-cipher";
+ channel_stats.ssl_cipher = "the-ssl-cipher";
cricket::TransportStats transport_stats;
transport_stats.content_name = "audio";
@@ -713,6 +715,18 @@
} else {
EXPECT_EQ(kNotFound, remote_certificate_id);
}
+
+ // Check negotiated ciphers.
+ std::string dtls_cipher = ExtractStatsValue(
+ StatsReport::kStatsReportTypeComponent,
+ reports,
+ StatsReport::kStatsValueNameDtlsCipher);
+ EXPECT_EQ("the-ssl-cipher", dtls_cipher);
+ std::string srtp_cipher = ExtractStatsValue(
+ StatsReport::kStatsReportTypeComponent,
+ reports,
+ StatsReport::kStatsValueNameSrtpCipher);
+ EXPECT_EQ("the-srtp-cipher", srtp_cipher);
}
cricket::FakeMediaEngine* media_engine_;
@@ -1318,6 +1332,18 @@
reports,
StatsReport::kStatsValueNameRemoteCertificateId);
ASSERT_EQ(kNotFound, remote_certificate_id);
+
+ // Check that the negotiated ciphers are absent.
+ std::string dtls_cipher = ExtractStatsValue(
+ StatsReport::kStatsReportTypeComponent,
+ reports,
+ StatsReport::kStatsValueNameDtlsCipher);
+ ASSERT_EQ(kNotFound, dtls_cipher);
+ std::string srtp_cipher = ExtractStatsValue(
+ StatsReport::kStatsReportTypeComponent,
+ reports,
+ StatsReport::kStatsValueNameSrtpCipher);
+ ASSERT_EQ(kNotFound, srtp_cipher);
}
// This test verifies that the stats are generated correctly when the transport
diff --git a/talk/app/webrtc/statstypes.cc b/talk/app/webrtc/statstypes.cc
index f41bf7a..e88583f 100644
--- a/talk/app/webrtc/statstypes.cc
+++ b/talk/app/webrtc/statstypes.cc
@@ -54,7 +54,7 @@
case StatsReport::kStatsReportTypeIceRemoteCandidate:
return "remotecandidate";
case StatsReport::kStatsReportTypeTransport:
- return "googTransport";
+ return "transport";
case StatsReport::kStatsReportTypeComponent:
return "googComponent";
case StatsReport::kStatsReportTypeCandidatePair:
@@ -244,6 +244,8 @@
return "protocol";
case kStatsValueNameTransportId:
return "transportId";
+ case kStatsValueNameSelectedCandidatePairId:
+ return "selectedCandidatePairId";
case kStatsValueNameSsrc:
return "ssrc";
case kStatsValueNameState:
@@ -310,6 +312,8 @@
return "googDecodingPLCCNG";
case kStatsValueNameDer:
return "googDerBase64";
+ case kStatsValueNameDtlsCipher:
+ return "dtlsCipher";
case kStatsValueNameEchoCancellationQualityMin:
return "googEchoCancellationQualityMin";
case kStatsValueNameEchoDelayMedian:
@@ -383,7 +387,7 @@
case kStatsValueNameLocalCandidateType:
return "googLocalCandidateType";
case kStatsValueNameLocalCertificateId:
- return "googLocalCertificateId";
+ return "localCertificateId";
case kStatsValueNameAdaptationChanges:
return "googAdaptationChanges";
case kStatsValueNameNacksReceived:
@@ -411,7 +415,7 @@
case kStatsValueNameRemoteCandidateType:
return "googRemoteCandidateType";
case kStatsValueNameRemoteCertificateId:
- return "googRemoteCertificateId";
+ return "remoteCertificateId";
case kStatsValueNameRetransmitBitrate:
return "googRetransmitBitrate";
case kStatsValueNameRtt:
@@ -422,6 +426,8 @@
return "packetsDiscardedOnSend";
case kStatsValueNameSpeechExpandRate:
return "googSpeechExpandRate";
+ case kStatsValueNameSrtpCipher:
+ return "srtpCipher";
case kStatsValueNameTargetEncBitrate:
return "googTargetEncBitrate";
case kStatsValueNameTransmitBitrate:
diff --git a/talk/app/webrtc/statstypes.h b/talk/app/webrtc/statstypes.h
index 65549df..9fe9c11 100644
--- a/talk/app/webrtc/statstypes.h
+++ b/talk/app/webrtc/statstypes.h
@@ -125,6 +125,7 @@
kStatsValueNamePacketsSent,
kStatsValueNameProtocol,
kStatsValueNameReadable,
+ kStatsValueNameSelectedCandidatePairId,
kStatsValueNameSsrc,
kStatsValueNameState,
kStatsValueNameTransportId,
@@ -160,6 +161,7 @@
kStatsValueNameDecodingPLC,
kStatsValueNameDecodingPLCCNG,
kStatsValueNameDer,
+ kStatsValueNameDtlsCipher,
kStatsValueNameEchoCancellationQualityMin,
kStatsValueNameEchoDelayMedian,
kStatsValueNameEchoDelayStdDev,
@@ -211,6 +213,7 @@
kStatsValueNameSecondaryDecodedRate,
kStatsValueNameSendPacketsDiscarded,
kStatsValueNameSpeechExpandRate,
+ kStatsValueNameSrtpCipher,
kStatsValueNameTargetDelayMs,
kStatsValueNameTargetEncBitrate,
kStatsValueNameTrackId,
diff --git a/talk/app/webrtc/test/mockpeerconnectionobservers.h b/talk/app/webrtc/test/mockpeerconnectionobservers.h
index 9b06c2f..4ab6e8e 100644
--- a/talk/app/webrtc/test/mockpeerconnectionobservers.h
+++ b/talk/app/webrtc/test/mockpeerconnectionobservers.h
@@ -123,7 +123,7 @@
virtual void OnComplete(const StatsReports& reports) {
ASSERT(!called_);
called_ = true;
- memset(&stats_, 0, sizeof(stats_));
+ stats_.Clear();
stats_.number_of_reports = reports.size();
for (const auto* r : reports) {
if (r->type() == StatsReport::kStatsReportTypeSsrc) {
@@ -138,6 +138,11 @@
} else if (r->type() == StatsReport::kStatsReportTypeBwe) {
GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
&stats_.available_receive_bandwidth);
+ } else if (r->type() == StatsReport::kStatsReportTypeComponent) {
+ GetStringValue(r, StatsReport::kStatsValueNameDtlsCipher,
+ &stats_.dtls_cipher);
+ GetStringValue(r, StatsReport::kStatsValueNameSrtpCipher,
+ &stats_.srtp_cipher);
}
}
}
@@ -170,6 +175,16 @@
return stats_.available_receive_bandwidth;
}
+ std::string DtlsCipher() const {
+ ASSERT(called_);
+ return stats_.dtls_cipher;
+ }
+
+ std::string SrtpCipher() const {
+ ASSERT(called_);
+ return stats_.srtp_cipher;
+ }
+
private:
bool GetIntValue(const StatsReport* report,
StatsReport::StatsValueName name,
@@ -182,15 +197,39 @@
}
return false;
}
+ bool GetStringValue(const StatsReport* report,
+ StatsReport::StatsValueName name,
+ std::string* value) {
+ for (const auto& v : report->values()) {
+ if (v->name == name) {
+ *value = v->value;
+ return true;
+ }
+ }
+ return false;
+ }
bool called_;
struct {
+ void Clear() {
+ number_of_reports = 0;
+ audio_output_level = 0;
+ audio_input_level = 0;
+ bytes_received = 0;
+ bytes_sent = 0;
+ available_receive_bandwidth = 0;
+ dtls_cipher.clear();
+ srtp_cipher.clear();
+ }
+
size_t number_of_reports;
int audio_output_level;
int audio_input_level;
int bytes_received;
int bytes_sent;
int available_receive_bandwidth;
+ std::string dtls_cipher;
+ std::string srtp_cipher;
} stats_;
};
