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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/rtc_base/ssl_stream_adapter_unittest.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/rtc_base/ssl_stream_adapter_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/rtc_base/ssl_stream_adapter_unittest.cc | 1853 |
1 files changed, 1853 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_base/ssl_stream_adapter_unittest.cc b/third_party/libwebrtc/rtc_base/ssl_stream_adapter_unittest.cc new file mode 100644 index 0000000000..0a99d9b1f0 --- /dev/null +++ b/third_party/libwebrtc/rtc_base/ssl_stream_adapter_unittest.cc @@ -0,0 +1,1853 @@ +/* + * Copyright 2011 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. + */ + +#include "rtc_base/ssl_stream_adapter.h" + +#include <algorithm> +#include <memory> +#include <set> +#include <string> + +#include "absl/memory/memory.h" +#include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "rtc_base/buffer_queue.h" +#include "rtc_base/checks.h" +#include "rtc_base/gunit.h" +#include "rtc_base/helpers.h" +#include "rtc_base/memory/fifo_buffer.h" +#include "rtc_base/memory_stream.h" +#include "rtc_base/message_digest.h" +#include "rtc_base/openssl_stream_adapter.h" +#include "rtc_base/ssl_adapter.h" +#include "rtc_base/ssl_identity.h" +#include "rtc_base/stream.h" +#include "test/field_trial.h" + +using ::testing::Combine; +using ::testing::tuple; +using ::testing::Values; +using ::testing::WithParamInterface; +using ::webrtc::SafeTask; + +static const int kBlockSize = 4096; +static const char kExporterLabel[] = "label"; +static const unsigned char kExporterContext[] = "context"; +static int kExporterContextLen = sizeof(kExporterContext); + +// A private key used for testing, broken into pieces in order to avoid +// issues with Git's checks for private keys in repos. +#define RSA_PRIVATE_KEY_HEADER "-----BEGIN RSA PRIVATE KEY-----\n" + +static const char kRSA_PRIVATE_KEY_PEM[] = RSA_PRIVATE_KEY_HEADER + "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n" + "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n" + "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n" + "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n" + "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n" + "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n" + "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n" + "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n" + "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n" + "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n" + "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n" + "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n" + "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n" + "UCXiYxSsu20QNVw=\n" + "-----END RSA PRIVATE KEY-----\n"; + +#undef RSA_PRIVATE_KEY_HEADER + +static const char kCERT_PEM[] = + "-----BEGIN CERTIFICATE-----\n" + "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n" + "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n" + "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n" + "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n" + "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n" + "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n" + "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n" + "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n" + "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n" + "-----END CERTIFICATE-----\n"; + +static const char kIntCert1[] = + "-----BEGIN CERTIFICATE-----\n" + "MIIEUjCCAjqgAwIBAgIBAjANBgkqhkiG9w0BAQsFADCBljELMAkGA1UEBhMCVVMx\n" + "EzARBgNVBAgMCkNhbGlmb3JuaWExFjAUBgNVBAcMDU1vdW50YWluIFZpZXcxFDAS\n" + "BgNVBAoMC0dvb2dsZSwgSW5jMQwwCgYDVQQLDANHVFAxFzAVBgNVBAMMDnRlbGVw\n" + "aG9ueS5nb29nMR0wGwYJKoZIhvcNAQkBFg5ndHBAZ29vZ2xlLmNvbTAeFw0xNzA5\n" + "MjYwNDA5MDNaFw0yMDA2MjIwNDA5MDNaMGQxCzAJBgNVBAYTAlVTMQswCQYDVQQI\n" + "DAJDQTEWMBQGA1UEBwwNTW91bnRhaW4gVmlldzEXMBUGA1UECgwOdGVsZXBob255\n" + "Lmdvb2cxFzAVBgNVBAMMDnRlbGVwaG9ueS5nb29nMIGfMA0GCSqGSIb3DQEBAQUA\n" + "A4GNADCBiQKBgQDJXWeeU1v1+wlqkVobzI3aN7Uh2iVQA9YCdq5suuabtiD/qoOD\n" + "NKpmQqsx7WZGGWSZTDFEBaUpvIK7Hb+nzRqk6iioPCFOFuarm6GxO1xVneImMuE6\n" + "tuWb3YZPr+ikChJbl11y5UcSbg0QsbeUc+jHl5umNvrL85Y+z8SP0rxbBwIDAQAB\n" + "o2AwXjAdBgNVHQ4EFgQU7tdZobqlN8R8V72FQnRxmqq8tKswHwYDVR0jBBgwFoAU\n" + "5GgKMUtcxkQ2dJrtNR5YOlIAPDswDwYDVR0TAQH/BAUwAwEB/zALBgNVHQ8EBAMC\n" + "AQYwDQYJKoZIhvcNAQELBQADggIBADObh9Z+z14FmP9zSenhFtq7hFnmNrSkklk8\n" + "eyYWXKfOuIriEQQBZsz76ZcnzStih8Rj+yQ0AXydk4fJ5LOwC2cUqQBar17g6Pd2\n" + "8g4SIL4azR9WvtiSvpuGlwp25b+yunaacDne6ebnf/MUiiKT5w61Xo3cEPVfl38e\n" + "/Up2l0bioid5enUTmg6LY6RxDO6tnZQkz3XD+nNSwT4ehtkqFpHYWjErj0BbkDM2\n" + "hiVc/JsYOZn3DmuOlHVHU6sKwqh3JEyvHO/d7DGzMGWHpHwv2mCTJq6l/sR95Tc2\n" + "GaQZgGDVNs9pdEouJCDm9e/PbQWRYhnat82PTkXx/6mDAAwdZlIi/pACzq8K4p7e\n" + "6hF0t8uKGnXJubHPXxlnJU6yxZ0yWmivAGjwWK4ur832gKlho4jeMDhiI/T3QPpl\n" + "iMNsIvxRhdD+GxJkQP1ezayw8s+Uc9KwKglrkBSRRDLCJUfPOvMmXLUDSTMX7kp4\n" + "/Ak1CA8dVLJIlfEjLBUuvAttlP7+7lsKNgxAjCxZkWLXIyGULzNPQwVWkGfCbrQs\n" + "XyMvSbFsSIb7blV7eLlmf9a+2RprUUkc2ALXLLCI9YQXmxm2beBfMyNmmebwBJzT\n" + "B0OR+5pFFNTJPoNlqpdrDsGrDu7JlUtk0ZLZzYyKXbgy2qXxfd4OWzXXjxpLMszZ\n" + "LDIpOAkj\n" + "-----END CERTIFICATE-----\n"; + +static const char kCACert[] = + "-----BEGIN CERTIFICATE-----\n" + "MIIGETCCA/mgAwIBAgIJAKN9r/BdbGUJMA0GCSqGSIb3DQEBCwUAMIGWMQswCQYD\n" + "VQQGEwJVUzETMBEGA1UECAwKQ2FsaWZvcm5pYTEWMBQGA1UEBwwNTW91bnRhaW4g\n" + "VmlldzEUMBIGA1UECgwLR29vZ2xlLCBJbmMxDDAKBgNVBAsMA0dUUDEXMBUGA1UE\n" + "AwwOdGVsZXBob255Lmdvb2cxHTAbBgkqhkiG9w0BCQEWDmd0cEBnb29nbGUuY29t\n" + "MB4XDTE3MDcyNzIzMDE0NVoXDTE3MDgyNjIzMDE0NVowgZYxCzAJBgNVBAYTAlVT\n" + "MRMwEQYDVQQIDApDYWxpZm9ybmlhMRYwFAYDVQQHDA1Nb3VudGFpbiBWaWV3MRQw\n" + "EgYDVQQKDAtHb29nbGUsIEluYzEMMAoGA1UECwwDR1RQMRcwFQYDVQQDDA50ZWxl\n" + "cGhvbnkuZ29vZzEdMBsGCSqGSIb3DQEJARYOZ3RwQGdvb2dsZS5jb20wggIiMA0G\n" + "CSqGSIb3DQEBAQUAA4ICDwAwggIKAoICAQCfvpF7aBV5Hp1EHsWoIlL3GeHwh8dS\n" + "lv9VQCegN9rD06Ny7MgcED5AiK2vqXmUmOVS+7NbATkdVYN/eozDhKtN3Q3n87kJ\n" + "Nt/TD/TcZZHOZIGsRPbrf2URK26E/5KzTzbzXVBOA1e+gSj+EBbltGqb01ZO5ErF\n" + "iPGViPM/HpYKdq6mfz2bS5PhU67XZMM2zvToyReQ/Fjm/6PJhwKSRXSgZF5djPhk\n" + "2LfOKMLS0AeZtd2C4DFsCU41lfLUkybioDgFuzTQ3TFi1K8A07KYTMmLY/yQppnf\n" + "SpNX58shlVhM+Ed37K1Z0rU0OfVCZ5P+KKaSSfMranjlU7zeUIhZYjqq/EYrEhbS\n" + "dLnNHwgJrqxzId3kq8uuLM6+VB7JZKnZLfT90GdAbX4+tutNe21smmogF9f80vEy\n" + "gM4tOp9rXrvz9vCwWHXVY9kdKemdLAsREoO6MS9k2ctK4jj80o2dROuFC6Q3e7mz\n" + "RjvZr5Tvi464c2o9o/jNlJ0O6q7V2eQzohD+7VnV5QPpRGXxlIeqpR2zoAg+WtRS\n" + "4OgHOVYiD3M6uAlggJA5pcDjMfkEZ+pkhtVcT4qMCEoruk6GbyPxS565oSHu16bH\n" + "EjeCqbZOVND5T3oA7nz6aQSs8sJabt0jmxUkGVnE+4ZDIuuRtkRma+0P/96Mtqor\n" + "OlpNWY1OBDY64QIDAQABo2AwXjAdBgNVHQ4EFgQU5GgKMUtcxkQ2dJrtNR5YOlIA\n" + "PDswHwYDVR0jBBgwFoAU5GgKMUtcxkQ2dJrtNR5YOlIAPDswDwYDVR0TAQH/BAUw\n" + "AwEB/zALBgNVHQ8EBAMCAQYwDQYJKoZIhvcNAQELBQADggIBAARQly5/bB6VUL2C\n" + "ykDYgWt48go407pAra6tL2kjpdfxV5PdL7iMZRkeht00vj+BVahIqZKrNOa/f5Fx\n" + "vlpahZFu0PDN436aQwRZ9qWut2qDOK0/z9Hhj6NWybquRFwMwqkPG/ivLMDU8Dmj\n" + "CIplpngPYNwXCs0KzdjSXYxqxJbwMjQXELD+/RcurY0oTtJMM1/2vKQMzw24UJqe\n" + "XLJAlsnd2AnWzWNUEviDZY89j9NdkHerBmV2gGzcU+X5lgOO5M8odBv0ZC9D+a6Z\n" + "QPZAOfdGVw60hhGvTW5s/s0dHwCpegRidhs0MD0fTmwwjYFBSmUx3Gztr4JTzOOr\n" + "7e5daJuak2ujQ5DqcGBvt1gePjSudb5brS7JQtN8tI/FyrnR4q/OuOwv1EvlC5RG\n" + "hLX+TXaWqFxB1Hd8ebKRR40mboFG6KcUI3lLBthDvQE7jnq48QfZMjlMQK0ZF1l7\n" + "SrlwRXWA74bU8CLJvnZKKo9p4TsTiDYGSYC6tNHKj5s3TGWL46oqGyZ0KdGNhrtC\n" + "rIGenMhth1vPYjyy0XuGBndXT85yi+IM2l8g8oU845+plxIhgpSI8bbC0oLwnhQ5\n" + "ARfsiYLkXDE7imSS0CSUmye76372mlzAIB1is4bBB/SzpPQtBuB9LDKtONgpSGHn\n" + "dGaXBy+qbVXVyGXaeEbIRjtJ6m92\n" + "-----END CERTIFICATE-----\n"; + +class SSLStreamAdapterTestBase; + +class SSLDummyStreamBase : public rtc::StreamInterface, + public sigslot::has_slots<> { + public: + SSLDummyStreamBase(SSLStreamAdapterTestBase* test, + absl::string_view side, + rtc::StreamInterface* in, + rtc::StreamInterface* out) + : test_base_(test), side_(side), in_(in), out_(out), first_packet_(true) { + in_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventIn); + out_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventOut); + } + + rtc::StreamState GetState() const override { return rtc::SS_OPEN; } + + rtc::StreamResult Read(rtc::ArrayView<uint8_t> buffer, + size_t& read, + int& error) override { + rtc::StreamResult r; + + r = in_->Read(buffer, read, error); + if (r == rtc::SR_BLOCK) + return rtc::SR_BLOCK; + if (r == rtc::SR_EOS) + return rtc::SR_EOS; + + if (r != rtc::SR_SUCCESS) { + ADD_FAILURE(); + return rtc::SR_ERROR; + } + + return rtc::SR_SUCCESS; + } + + // Catch readability events on in and pass them up. + void OnEventIn(rtc::StreamInterface* stream, int sig, int err) { + int mask = (rtc::SE_READ | rtc::SE_CLOSE); + + if (sig & mask) { + RTC_LOG(LS_VERBOSE) << "SSLDummyStreamBase::OnEvent side=" << side_ + << " sig=" << sig << " forwarding upward"; + PostEvent(sig & mask, 0); + } + } + + // Catch writeability events on out and pass them up. + void OnEventOut(rtc::StreamInterface* stream, int sig, int err) { + if (sig & rtc::SE_WRITE) { + RTC_LOG(LS_VERBOSE) << "SSLDummyStreamBase::OnEvent side=" << side_ + << " sig=" << sig << " forwarding upward"; + + PostEvent(sig & rtc::SE_WRITE, 0); + } + } + + // Write to the outgoing FifoBuffer + rtc::StreamResult WriteData(rtc::ArrayView<const uint8_t> data, + size_t& written, + int& error) { + return out_->Write(data, written, error); + } + + rtc::StreamResult Write(rtc::ArrayView<const uint8_t> data, + size_t& written, + int& error) override; + + void Close() override { + RTC_LOG(LS_INFO) << "Closing outbound stream"; + out_->Close(); + } + + private: + void PostEvent(int events, int err) { + thread_->PostTask(SafeTask(task_safety_.flag(), [this, events, err]() { + SignalEvent(this, events, err); + })); + } + + webrtc::ScopedTaskSafety task_safety_; + rtc::Thread* const thread_ = rtc::Thread::Current(); + SSLStreamAdapterTestBase* test_base_; + const std::string side_; + rtc::StreamInterface* in_; + rtc::StreamInterface* out_; + bool first_packet_; +}; + +class SSLDummyStreamTLS : public SSLDummyStreamBase { + public: + SSLDummyStreamTLS(SSLStreamAdapterTestBase* test, + absl::string_view side, + rtc::FifoBuffer* in, + rtc::FifoBuffer* out) + : SSLDummyStreamBase(test, side, in, out) {} +}; + +class BufferQueueStream : public rtc::StreamInterface { + public: + BufferQueueStream(size_t capacity, size_t default_size) + : buffer_(capacity, default_size) {} + + // Implementation of abstract StreamInterface methods. + + // A buffer queue stream is always "open". + rtc::StreamState GetState() const override { return rtc::SS_OPEN; } + + // Reading a buffer queue stream will either succeed or block. + rtc::StreamResult Read(rtc::ArrayView<uint8_t> buffer, + size_t& read, + int& error) override { + const bool was_writable = buffer_.is_writable(); + if (!buffer_.ReadFront(buffer.data(), buffer.size(), &read)) + return rtc::SR_BLOCK; + + if (!was_writable) + NotifyWritableForTest(); + + return rtc::SR_SUCCESS; + } + + // Writing to a buffer queue stream will either succeed or block. + rtc::StreamResult Write(rtc::ArrayView<const uint8_t> data, + size_t& written, + int& error) override { + const bool was_readable = buffer_.is_readable(); + if (!buffer_.WriteBack(data.data(), data.size(), &written)) + return rtc::SR_BLOCK; + + if (!was_readable) + NotifyReadableForTest(); + + return rtc::SR_SUCCESS; + } + + // A buffer queue stream can not be closed. + void Close() override {} + + protected: + void NotifyReadableForTest() { PostEvent(rtc::SE_READ, 0); } + void NotifyWritableForTest() { PostEvent(rtc::SE_WRITE, 0); } + + private: + void PostEvent(int events, int err) { + thread_->PostTask(SafeTask(task_safety_.flag(), [this, events, err]() { + SignalEvent(this, events, err); + })); + } + + rtc::Thread* const thread_ = rtc::Thread::Current(); + webrtc::ScopedTaskSafety task_safety_; + rtc::BufferQueue buffer_; +}; + +class SSLDummyStreamDTLS : public SSLDummyStreamBase { + public: + SSLDummyStreamDTLS(SSLStreamAdapterTestBase* test, + absl::string_view side, + BufferQueueStream* in, + BufferQueueStream* out) + : SSLDummyStreamBase(test, side, in, out) {} +}; + +static const int kFifoBufferSize = 4096; +static const int kBufferCapacity = 1; +static const size_t kDefaultBufferSize = 2048; + +class SSLStreamAdapterTestBase : public ::testing::Test, + public sigslot::has_slots<> { + public: + SSLStreamAdapterTestBase( + absl::string_view client_cert_pem, + absl::string_view client_private_key_pem, + bool dtls, + rtc::KeyParams client_key_type = rtc::KeyParams(rtc::KT_DEFAULT), + rtc::KeyParams server_key_type = rtc::KeyParams(rtc::KT_DEFAULT)) + : client_cert_pem_(client_cert_pem), + client_private_key_pem_(client_private_key_pem), + client_key_type_(client_key_type), + server_key_type_(server_key_type), + client_stream_(nullptr), + server_stream_(nullptr), + delay_(0), + mtu_(1460), + loss_(0), + lose_first_packet_(false), + damage_(false), + dtls_(dtls), + handshake_wait_(5000), + identities_set_(false) { + // Set use of the test RNG to get predictable loss patterns. + rtc::SetRandomTestMode(true); + } + + ~SSLStreamAdapterTestBase() override { + // Put it back for the next test. + rtc::SetRandomTestMode(false); + } + + void SetUp() override { + CreateStreams(); + + client_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(client_stream_)); + server_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(server_stream_)); + + // Set up the slots + client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent); + server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent); + + std::unique_ptr<rtc::SSLIdentity> client_identity; + if (!client_cert_pem_.empty() && !client_private_key_pem_.empty()) { + client_identity = rtc::SSLIdentity::CreateFromPEMStrings( + client_private_key_pem_, client_cert_pem_); + } else { + client_identity = rtc::SSLIdentity::Create("client", client_key_type_); + } + auto server_identity = rtc::SSLIdentity::Create("server", server_key_type_); + + client_ssl_->SetIdentity(std::move(client_identity)); + server_ssl_->SetIdentity(std::move(server_identity)); + } + + void TearDown() override { + client_ssl_.reset(nullptr); + server_ssl_.reset(nullptr); + } + + virtual void CreateStreams() = 0; + + // Recreate the client/server identities with the specified validity period. + // `not_before` and `not_after` are offsets from the current time in number + // of seconds. + void ResetIdentitiesWithValidity(int not_before, int not_after) { + CreateStreams(); + + client_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(client_stream_)); + server_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(server_stream_)); + + client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent); + server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent); + + time_t now = time(nullptr); + + rtc::SSLIdentityParams client_params; + client_params.key_params = rtc::KeyParams(rtc::KT_DEFAULT); + client_params.common_name = "client"; + client_params.not_before = now + not_before; + client_params.not_after = now + not_after; + auto client_identity = rtc::SSLIdentity::CreateForTest(client_params); + + rtc::SSLIdentityParams server_params; + server_params.key_params = rtc::KeyParams(rtc::KT_DEFAULT); + server_params.common_name = "server"; + server_params.not_before = now + not_before; + server_params.not_after = now + not_after; + auto server_identity = rtc::SSLIdentity::CreateForTest(server_params); + + client_ssl_->SetIdentity(std::move(client_identity)); + server_ssl_->SetIdentity(std::move(server_identity)); + } + + virtual void OnEvent(rtc::StreamInterface* stream, int sig, int err) { + RTC_LOG(LS_VERBOSE) << "SSLStreamAdapterTestBase::OnEvent sig=" << sig; + + if (sig & rtc::SE_READ) { + ReadData(stream); + } + + if ((stream == client_ssl_.get()) && (sig & rtc::SE_WRITE)) { + WriteData(); + } + } + + void SetPeerIdentitiesByDigest(bool correct, bool expect_success) { + unsigned char server_digest[20]; + size_t server_digest_len; + unsigned char client_digest[20]; + size_t client_digest_len; + bool rv; + rtc::SSLPeerCertificateDigestError err; + rtc::SSLPeerCertificateDigestError expected_err = + expect_success + ? rtc::SSLPeerCertificateDigestError::NONE + : rtc::SSLPeerCertificateDigestError::VERIFICATION_FAILED; + + RTC_LOG(LS_INFO) << "Setting peer identities by digest"; + + rv = server_identity()->certificate().ComputeDigest( + rtc::DIGEST_SHA_1, server_digest, 20, &server_digest_len); + ASSERT_TRUE(rv); + rv = client_identity()->certificate().ComputeDigest( + rtc::DIGEST_SHA_1, client_digest, 20, &client_digest_len); + ASSERT_TRUE(rv); + + if (!correct) { + RTC_LOG(LS_INFO) << "Setting bogus digest for server cert"; + server_digest[0]++; + } + rv = client_ssl_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, server_digest, + server_digest_len, &err); + EXPECT_EQ(expected_err, err); + EXPECT_EQ(expect_success, rv); + + if (!correct) { + RTC_LOG(LS_INFO) << "Setting bogus digest for client cert"; + client_digest[0]++; + } + rv = server_ssl_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, client_digest, + client_digest_len, &err); + EXPECT_EQ(expected_err, err); + EXPECT_EQ(expect_success, rv); + + identities_set_ = true; + } + + void SetupProtocolVersions(rtc::SSLProtocolVersion server_version, + rtc::SSLProtocolVersion client_version) { + server_ssl_->SetMaxProtocolVersion(server_version); + client_ssl_->SetMaxProtocolVersion(client_version); + } + + void TestHandshake(bool expect_success = true) { + server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS); + client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS); + + if (!dtls_) { + // Make sure we simulate a reliable network for TLS. + // This is just a check to make sure that people don't write wrong + // tests. + RTC_CHECK_EQ(1460, mtu_); + RTC_CHECK(!loss_); + RTC_CHECK(!lose_first_packet_); + } + + if (!identities_set_) + SetPeerIdentitiesByDigest(true, true); + + // Start the handshake + int rv; + + server_ssl_->SetServerRole(); + rv = server_ssl_->StartSSL(); + ASSERT_EQ(0, rv); + + rv = client_ssl_->StartSSL(); + ASSERT_EQ(0, rv); + + // Now run the handshake + if (expect_success) { + EXPECT_TRUE_WAIT((client_ssl_->GetState() == rtc::SS_OPEN) && + (server_ssl_->GetState() == rtc::SS_OPEN), + handshake_wait_); + } else { + EXPECT_TRUE_WAIT(client_ssl_->GetState() == rtc::SS_CLOSED, + handshake_wait_); + } + } + + // This tests that we give up after 12 DTLS resends. + void TestHandshakeTimeout() { + rtc::ScopedFakeClock clock; + int64_t time_start = clock.TimeNanos(); + webrtc::TimeDelta time_increment = webrtc::TimeDelta::Millis(1000); + server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS); + client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS); + + if (!dtls_) { + // Make sure we simulate a reliable network for TLS. + // This is just a check to make sure that people don't write wrong + // tests. + RTC_CHECK_EQ(1460, mtu_); + RTC_CHECK(!loss_); + RTC_CHECK(!lose_first_packet_); + } + + if (!identities_set_) + SetPeerIdentitiesByDigest(true, true); + + // Start the handshake + int rv; + + server_ssl_->SetServerRole(); + rv = server_ssl_->StartSSL(); + ASSERT_EQ(0, rv); + + rv = client_ssl_->StartSSL(); + ASSERT_EQ(0, rv); + + // Now wait for the handshake to timeout (or fail after an hour of simulated + // time). + while (client_ssl_->GetState() == rtc::SS_OPENING && + (rtc::TimeDiff(clock.TimeNanos(), time_start) < + 3600 * rtc::kNumNanosecsPerSec)) { + EXPECT_TRUE_WAIT(!((client_ssl_->GetState() == rtc::SS_OPEN) && + (server_ssl_->GetState() == rtc::SS_OPEN)), + 1000); + clock.AdvanceTime(time_increment); + } + RTC_CHECK_EQ(client_ssl_->GetState(), rtc::SS_CLOSED); + } + + // This tests that the handshake can complete before the identity is verified, + // and the identity will be verified after the fact. It also verifies that + // packets can't be read or written before the identity has been verified. + void TestHandshakeWithDelayedIdentity(bool valid_identity) { + server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS); + client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS); + + if (!dtls_) { + // Make sure we simulate a reliable network for TLS. + // This is just a check to make sure that people don't write wrong + // tests. + RTC_CHECK_EQ(1460, mtu_); + RTC_CHECK(!loss_); + RTC_CHECK(!lose_first_packet_); + } + + // Start the handshake + server_ssl_->SetServerRole(); + ASSERT_EQ(0, server_ssl_->StartSSL()); + ASSERT_EQ(0, client_ssl_->StartSSL()); + + // Now run the handshake. + EXPECT_TRUE_WAIT( + client_ssl_->IsTlsConnected() && server_ssl_->IsTlsConnected(), + handshake_wait_); + + // Until the identity has been verified, the state should still be + // SS_OPENING and writes should return SR_BLOCK. + EXPECT_EQ(rtc::SS_OPENING, client_ssl_->GetState()); + EXPECT_EQ(rtc::SS_OPENING, server_ssl_->GetState()); + uint8_t packet[1]; + size_t sent; + int error; + EXPECT_EQ(rtc::SR_BLOCK, client_ssl_->Write(packet, sent, error)); + EXPECT_EQ(rtc::SR_BLOCK, server_ssl_->Write(packet, sent, error)); + + // Collect both of the certificate digests; needs to be done before calling + // SetPeerCertificateDigest as that may reset the identity. + unsigned char server_digest[20]; + size_t server_digest_len; + unsigned char client_digest[20]; + size_t client_digest_len; + bool rv; + + rv = server_identity()->certificate().ComputeDigest( + rtc::DIGEST_SHA_1, server_digest, 20, &server_digest_len); + ASSERT_TRUE(rv); + rv = client_identity()->certificate().ComputeDigest( + rtc::DIGEST_SHA_1, client_digest, 20, &client_digest_len); + ASSERT_TRUE(rv); + + if (!valid_identity) { + RTC_LOG(LS_INFO) << "Setting bogus digest for client/server certs"; + client_digest[0]++; + server_digest[0]++; + } + + // Set the peer certificate digest for the client. + rtc::SSLPeerCertificateDigestError err; + rtc::SSLPeerCertificateDigestError expected_err = + valid_identity + ? rtc::SSLPeerCertificateDigestError::NONE + : rtc::SSLPeerCertificateDigestError::VERIFICATION_FAILED; + rv = client_ssl_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, server_digest, + server_digest_len, &err); + EXPECT_EQ(expected_err, err); + EXPECT_EQ(valid_identity, rv); + // State should then transition to SS_OPEN or SS_CLOSED based on validation + // of the identity. + if (valid_identity) { + EXPECT_EQ(rtc::SS_OPEN, client_ssl_->GetState()); + // If the client sends a packet while the server still hasn't verified the + // client identity, the server should continue to return SR_BLOCK. + int error; + EXPECT_EQ(rtc::SR_SUCCESS, client_ssl_->Write(packet, sent, error)); + size_t read; + EXPECT_EQ(rtc::SR_BLOCK, server_ssl_->Read(packet, read, error)); + } else { + EXPECT_EQ(rtc::SS_CLOSED, client_ssl_->GetState()); + } + + // Set the peer certificate digest for the server. + rv = server_ssl_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, client_digest, + client_digest_len, &err); + EXPECT_EQ(expected_err, err); + EXPECT_EQ(valid_identity, rv); + if (valid_identity) { + EXPECT_EQ(rtc::SS_OPEN, server_ssl_->GetState()); + } else { + EXPECT_EQ(rtc::SS_CLOSED, server_ssl_->GetState()); + } + } + + rtc::StreamResult DataWritten(SSLDummyStreamBase* from, + const void* data, + size_t data_len, + size_t& written, + int& error) { + // Randomly drop loss_ percent of packets + if (rtc::CreateRandomId() % 100 < static_cast<uint32_t>(loss_)) { + RTC_LOG(LS_VERBOSE) << "Randomly dropping packet, size=" << data_len; + written = data_len; + return rtc::SR_SUCCESS; + } + if (dtls_ && (data_len > mtu_)) { + RTC_LOG(LS_VERBOSE) << "Dropping packet > mtu, size=" << data_len; + written = data_len; + return rtc::SR_SUCCESS; + } + + // Optionally damage application data (type 23). Note that we don't damage + // handshake packets and we damage the last byte to keep the header + // intact but break the MAC. + if (damage_ && (*static_cast<const unsigned char*>(data) == 23)) { + std::vector<uint8_t> buf(data_len); + + RTC_LOG(LS_VERBOSE) << "Damaging packet"; + + memcpy(&buf[0], data, data_len); + buf[data_len - 1]++; + return from->WriteData(rtc::MakeArrayView(&buf[0], data_len), written, + error); + } + + return from->WriteData( + rtc::MakeArrayView(reinterpret_cast<const uint8_t*>(data), data_len), + written, error); + } + + void SetDelay(int delay) { delay_ = delay; } + int GetDelay() { return delay_; } + + void SetLoseFirstPacket(bool lose) { lose_first_packet_ = lose; } + bool GetLoseFirstPacket() { return lose_first_packet_; } + + void SetLoss(int percent) { loss_ = percent; } + + void SetDamage() { damage_ = true; } + + void SetMtu(size_t mtu) { mtu_ = mtu; } + + void SetHandshakeWait(int wait) { handshake_wait_ = wait; } + + void SetDtlsSrtpCryptoSuites(const std::vector<int>& ciphers, bool client) { + if (client) + client_ssl_->SetDtlsSrtpCryptoSuites(ciphers); + else + server_ssl_->SetDtlsSrtpCryptoSuites(ciphers); + } + + bool GetDtlsSrtpCryptoSuite(bool client, int* retval) { + if (client) + return client_ssl_->GetDtlsSrtpCryptoSuite(retval); + else + return server_ssl_->GetDtlsSrtpCryptoSuite(retval); + } + + std::unique_ptr<rtc::SSLCertificate> GetPeerCertificate(bool client) { + std::unique_ptr<rtc::SSLCertChain> chain; + if (client) + chain = client_ssl_->GetPeerSSLCertChain(); + else + chain = server_ssl_->GetPeerSSLCertChain(); + return (chain && chain->GetSize()) ? chain->Get(0).Clone() : nullptr; + } + + bool GetSslCipherSuite(bool client, int* retval) { + if (client) + return client_ssl_->GetSslCipherSuite(retval); + else + return server_ssl_->GetSslCipherSuite(retval); + } + + int GetSslVersion(bool client) { + if (client) + return client_ssl_->GetSslVersion(); + else + return server_ssl_->GetSslVersion(); + } + + bool ExportKeyingMaterial(absl::string_view label, + const unsigned char* context, + size_t context_len, + bool use_context, + bool client, + unsigned char* result, + size_t result_len) { + if (client) + return client_ssl_->ExportKeyingMaterial(label, context, context_len, + use_context, result, result_len); + else + return server_ssl_->ExportKeyingMaterial(label, context, context_len, + use_context, result, result_len); + } + + // To be implemented by subclasses. + virtual void WriteData() = 0; + virtual void ReadData(rtc::StreamInterface* stream) = 0; + virtual void TestTransfer(int size) = 0; + + protected: + rtc::SSLIdentity* client_identity() const { + if (!client_ssl_) { + return nullptr; + } + return client_ssl_->GetIdentityForTesting(); + } + rtc::SSLIdentity* server_identity() const { + if (!server_ssl_) { + return nullptr; + } + return server_ssl_->GetIdentityForTesting(); + } + + rtc::AutoThread main_thread_; + std::string client_cert_pem_; + std::string client_private_key_pem_; + rtc::KeyParams client_key_type_; + rtc::KeyParams server_key_type_; + SSLDummyStreamBase* client_stream_; // freed by client_ssl_ destructor + SSLDummyStreamBase* server_stream_; // freed by server_ssl_ destructor + std::unique_ptr<rtc::SSLStreamAdapter> client_ssl_; + std::unique_ptr<rtc::SSLStreamAdapter> server_ssl_; + int delay_; + size_t mtu_; + int loss_; + bool lose_first_packet_; + bool damage_; + bool dtls_; + int handshake_wait_; + bool identities_set_; +}; + +class SSLStreamAdapterTestTLS + : public SSLStreamAdapterTestBase, + public WithParamInterface<tuple<rtc::KeyParams, rtc::KeyParams>> { + public: + SSLStreamAdapterTestTLS() + : SSLStreamAdapterTestBase("", + "", + false, + ::testing::get<0>(GetParam()), + ::testing::get<1>(GetParam())), + client_buffer_(kFifoBufferSize), + server_buffer_(kFifoBufferSize) {} + + void CreateStreams() override { + client_stream_ = + new SSLDummyStreamTLS(this, "c2s", &client_buffer_, &server_buffer_); + server_stream_ = + new SSLDummyStreamTLS(this, "s2c", &server_buffer_, &client_buffer_); + } + + // Test data transfer for TLS + void TestTransfer(int size) override { + RTC_LOG(LS_INFO) << "Starting transfer test with " << size << " bytes"; + // Create some dummy data to send. + size_t received; + + send_stream_.ReserveSize(size); + for (int i = 0; i < size; ++i) { + uint8_t ch = static_cast<uint8_t>(i); + size_t written; + int error; + send_stream_.Write(rtc::MakeArrayView(&ch, 1), written, error); + } + send_stream_.Rewind(); + + // Prepare the receive stream. + recv_stream_.ReserveSize(size); + + // Start sending + WriteData(); + + // Wait for the client to close + EXPECT_TRUE_WAIT(server_ssl_->GetState() == rtc::SS_CLOSED, 10000); + + // Now check the data + recv_stream_.GetSize(&received); + + EXPECT_EQ(static_cast<size_t>(size), received); + EXPECT_EQ(0, + memcmp(send_stream_.GetBuffer(), recv_stream_.GetBuffer(), size)); + } + + void WriteData() override { + size_t position, tosend, size; + rtc::StreamResult rv; + size_t sent; + uint8_t block[kBlockSize]; + + send_stream_.GetSize(&size); + if (!size) + return; + + for (;;) { + send_stream_.GetPosition(&position); + int dummy_error; + if (send_stream_.Read(block, tosend, dummy_error) != rtc::SR_EOS) { + int error; + rv = client_ssl_->Write(rtc::MakeArrayView(block, tosend), sent, error); + + if (rv == rtc::SR_SUCCESS) { + send_stream_.SetPosition(position + sent); + RTC_LOG(LS_VERBOSE) << "Sent: " << position + sent; + } else if (rv == rtc::SR_BLOCK) { + RTC_LOG(LS_VERBOSE) << "Blocked..."; + send_stream_.SetPosition(position); + break; + } else { + ADD_FAILURE(); + break; + } + } else { + // Now close + RTC_LOG(LS_INFO) << "Wrote " << position << " bytes. Closing"; + client_ssl_->Close(); + break; + } + } + } + + void ReadData(rtc::StreamInterface* stream) override { + uint8_t buffer[1600]; + size_t bread; + int err2; + rtc::StreamResult r; + + for (;;) { + r = stream->Read(buffer, bread, err2); + + if (r == rtc::SR_ERROR || r == rtc::SR_EOS) { + // Unfortunately, errors are the way that the stream adapter + // signals close in OpenSSL. + stream->Close(); + return; + } + + if (r == rtc::SR_BLOCK) + break; + + ASSERT_EQ(rtc::SR_SUCCESS, r); + RTC_LOG(LS_VERBOSE) << "Read " << bread; + size_t written; + int error; + recv_stream_.Write(rtc::MakeArrayView(buffer, bread), written, error); + } + } + + private: + rtc::FifoBuffer client_buffer_; + rtc::FifoBuffer server_buffer_; + rtc::MemoryStream send_stream_; + rtc::MemoryStream recv_stream_; +}; + +class SSLStreamAdapterTestDTLSBase : public SSLStreamAdapterTestBase { + public: + SSLStreamAdapterTestDTLSBase(rtc::KeyParams param1, rtc::KeyParams param2) + : SSLStreamAdapterTestBase("", "", true, param1, param2), + client_buffer_(kBufferCapacity, kDefaultBufferSize), + server_buffer_(kBufferCapacity, kDefaultBufferSize), + packet_size_(1000), + count_(0), + sent_(0) {} + + SSLStreamAdapterTestDTLSBase(absl::string_view cert_pem, + absl::string_view private_key_pem) + : SSLStreamAdapterTestBase(cert_pem, private_key_pem, true), + client_buffer_(kBufferCapacity, kDefaultBufferSize), + server_buffer_(kBufferCapacity, kDefaultBufferSize), + packet_size_(1000), + count_(0), + sent_(0) {} + + void CreateStreams() override { + client_stream_ = + new SSLDummyStreamDTLS(this, "c2s", &client_buffer_, &server_buffer_); + server_stream_ = + new SSLDummyStreamDTLS(this, "s2c", &server_buffer_, &client_buffer_); + } + + void WriteData() override { + uint8_t* packet = new uint8_t[1600]; + + while (sent_ < count_) { + unsigned int rand_state = sent_; + packet[0] = sent_; + for (size_t i = 1; i < packet_size_; i++) { + // This is a simple LC PRNG. Keep in synch with identical code below. + rand_state = (rand_state * 251 + 19937) >> 7; + packet[i] = rand_state & 0xff; + } + + size_t sent; + int error; + rtc::StreamResult rv = client_ssl_->Write( + rtc::MakeArrayView(packet, packet_size_), sent, error); + if (rv == rtc::SR_SUCCESS) { + RTC_LOG(LS_VERBOSE) << "Sent: " << sent_; + sent_++; + } else if (rv == rtc::SR_BLOCK) { + RTC_LOG(LS_VERBOSE) << "Blocked..."; + break; + } else { + ADD_FAILURE(); + break; + } + } + + delete[] packet; + } + + void ReadData(rtc::StreamInterface* stream) override { + uint8_t buffer[2000]; + size_t bread; + int err2; + rtc::StreamResult r; + + for (;;) { + r = stream->Read(buffer, bread, err2); + + if (r == rtc::SR_ERROR) { + // Unfortunately, errors are the way that the stream adapter + // signals close right now + stream->Close(); + return; + } + + if (r == rtc::SR_BLOCK) + break; + + ASSERT_EQ(rtc::SR_SUCCESS, r); + RTC_LOG(LS_VERBOSE) << "Read " << bread; + + // Now parse the datagram + ASSERT_EQ(packet_size_, bread); + unsigned char packet_num = buffer[0]; + + unsigned int rand_state = packet_num; + for (size_t i = 1; i < packet_size_; i++) { + // This is a simple LC PRNG. Keep in synch with identical code above. + rand_state = (rand_state * 251 + 19937) >> 7; + ASSERT_EQ(rand_state & 0xff, buffer[i]); + } + received_.insert(packet_num); + } + } + + void TestTransfer(int count) override { + count_ = count; + + WriteData(); + + EXPECT_TRUE_WAIT(sent_ == count_, 10000); + RTC_LOG(LS_INFO) << "sent_ == " << sent_; + + if (damage_) { + WAIT(false, 2000); + EXPECT_EQ(0U, received_.size()); + } else if (loss_ == 0) { + EXPECT_EQ_WAIT(static_cast<size_t>(sent_), received_.size(), 1000); + } else { + RTC_LOG(LS_INFO) << "Sent " << sent_ << " packets; received " + << received_.size(); + } + } + + protected: + BufferQueueStream client_buffer_; + BufferQueueStream server_buffer_; + + private: + size_t packet_size_; + int count_; + int sent_; + std::set<int> received_; +}; + +class SSLStreamAdapterTestDTLS + : public SSLStreamAdapterTestDTLSBase, + public WithParamInterface<tuple<rtc::KeyParams, rtc::KeyParams>> { + public: + SSLStreamAdapterTestDTLS() + : SSLStreamAdapterTestDTLSBase(::testing::get<0>(GetParam()), + ::testing::get<1>(GetParam())) {} + + SSLStreamAdapterTestDTLS(absl::string_view cert_pem, + absl::string_view private_key_pem) + : SSLStreamAdapterTestDTLSBase(cert_pem, private_key_pem) {} +}; + +rtc::StreamResult SSLDummyStreamBase::Write(rtc::ArrayView<const uint8_t> data, + size_t& written, + int& error) { + RTC_LOG(LS_VERBOSE) << "Writing to loopback " << data.size(); + + if (first_packet_) { + first_packet_ = false; + if (test_base_->GetLoseFirstPacket()) { + RTC_LOG(LS_INFO) << "Losing initial packet of length " << data.size(); + written = data.size(); // Fake successful writing also to writer. + return rtc::SR_SUCCESS; + } + } + + return test_base_->DataWritten(this, data.data(), data.size(), written, + error); +} + +class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS { + public: + SSLStreamAdapterTestDTLSFromPEMStrings() + : SSLStreamAdapterTestDTLS(kCERT_PEM, kRSA_PRIVATE_KEY_PEM) {} +}; + +// Test fixture for certificate chaining. Server will push more than one +// certificate. +class SSLStreamAdapterTestDTLSCertChain : public SSLStreamAdapterTestDTLS { + public: + SSLStreamAdapterTestDTLSCertChain() : SSLStreamAdapterTestDTLS("", "") {} + void SetUp() override { + CreateStreams(); + + client_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(client_stream_)); + server_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(server_stream_)); + + // Set up the slots + client_ssl_->SignalEvent.connect( + reinterpret_cast<SSLStreamAdapterTestBase*>(this), + &SSLStreamAdapterTestBase::OnEvent); + server_ssl_->SignalEvent.connect( + reinterpret_cast<SSLStreamAdapterTestBase*>(this), + &SSLStreamAdapterTestBase::OnEvent); + + std::unique_ptr<rtc::SSLIdentity> client_identity; + if (!client_cert_pem_.empty() && !client_private_key_pem_.empty()) { + client_identity = rtc::SSLIdentity::CreateFromPEMStrings( + client_private_key_pem_, client_cert_pem_); + } else { + client_identity = rtc::SSLIdentity::Create("client", client_key_type_); + } + + client_ssl_->SetIdentity(std::move(client_identity)); + } +}; + +// Basic tests: TLS + +// Test that we can make a handshake work +TEST_P(SSLStreamAdapterTestTLS, TestTLSConnect) { + TestHandshake(); +} + +TEST_P(SSLStreamAdapterTestTLS, GetPeerCertChainWithOneCertificate) { + TestHandshake(); + std::unique_ptr<rtc::SSLCertChain> cert_chain = + client_ssl_->GetPeerSSLCertChain(); + ASSERT_NE(nullptr, cert_chain); + EXPECT_EQ(1u, cert_chain->GetSize()); + EXPECT_EQ(cert_chain->Get(0).ToPEMString(), + server_identity()->certificate().ToPEMString()); +} + +TEST_F(SSLStreamAdapterTestDTLSCertChain, TwoCertHandshake) { + auto server_identity = rtc::SSLIdentity::CreateFromPEMChainStrings( + kRSA_PRIVATE_KEY_PEM, std::string(kCERT_PEM) + kCACert); + server_ssl_->SetIdentity(std::move(server_identity)); + TestHandshake(); + std::unique_ptr<rtc::SSLCertChain> peer_cert_chain = + client_ssl_->GetPeerSSLCertChain(); + ASSERT_NE(nullptr, peer_cert_chain); + EXPECT_EQ(kCERT_PEM, peer_cert_chain->Get(0).ToPEMString()); + // TODO(bugs.webrtc.org/15153): Fix peer_cert_chain to return multiple + // certificates under OpenSSL. Today it only works with BoringSSL. +#ifdef OPENSSL_IS_BORINGSSL + ASSERT_EQ(2u, peer_cert_chain->GetSize()); + EXPECT_EQ(kCACert, peer_cert_chain->Get(1).ToPEMString()); +#endif +} + +TEST_F(SSLStreamAdapterTestDTLSCertChain, TwoCertHandshakeWithCopy) { + server_ssl_->SetIdentity(rtc::SSLIdentity::CreateFromPEMChainStrings( + kRSA_PRIVATE_KEY_PEM, std::string(kCERT_PEM) + kCACert)); + TestHandshake(); + std::unique_ptr<rtc::SSLCertChain> peer_cert_chain = + client_ssl_->GetPeerSSLCertChain(); + ASSERT_NE(nullptr, peer_cert_chain); + EXPECT_EQ(kCERT_PEM, peer_cert_chain->Get(0).ToPEMString()); + // TODO(bugs.webrtc.org/15153): Fix peer_cert_chain to return multiple + // certificates under OpenSSL. Today it only works with BoringSSL. +#ifdef OPENSSL_IS_BORINGSSL + ASSERT_EQ(2u, peer_cert_chain->GetSize()); + EXPECT_EQ(kCACert, peer_cert_chain->Get(1).ToPEMString()); +#endif +} + +TEST_F(SSLStreamAdapterTestDTLSCertChain, ThreeCertHandshake) { + server_ssl_->SetIdentity(rtc::SSLIdentity::CreateFromPEMChainStrings( + kRSA_PRIVATE_KEY_PEM, std::string(kCERT_PEM) + kIntCert1 + kCACert)); + TestHandshake(); + std::unique_ptr<rtc::SSLCertChain> peer_cert_chain = + client_ssl_->GetPeerSSLCertChain(); + ASSERT_NE(nullptr, peer_cert_chain); + EXPECT_EQ(kCERT_PEM, peer_cert_chain->Get(0).ToPEMString()); + // TODO(bugs.webrtc.org/15153): Fix peer_cert_chain to return multiple + // certificates under OpenSSL. Today it only works with BoringSSL. +#ifdef OPENSSL_IS_BORINGSSL + ASSERT_EQ(3u, peer_cert_chain->GetSize()); + EXPECT_EQ(kIntCert1, peer_cert_chain->Get(1).ToPEMString()); + EXPECT_EQ(kCACert, peer_cert_chain->Get(2).ToPEMString()); +#endif +} + +// Test that closing the connection on one side updates the other side. +TEST_P(SSLStreamAdapterTestTLS, TestTLSClose) { + TestHandshake(); + client_ssl_->Close(); + EXPECT_EQ_WAIT(rtc::SS_CLOSED, server_ssl_->GetState(), handshake_wait_); +} + +// Test transfer -- trivial +TEST_P(SSLStreamAdapterTestTLS, TestTLSTransfer) { + TestHandshake(); + TestTransfer(100000); +} + +// Test read-write after close. +TEST_P(SSLStreamAdapterTestTLS, ReadWriteAfterClose) { + TestHandshake(); + TestTransfer(100000); + client_ssl_->Close(); + + rtc::StreamResult rv; + uint8_t block[kBlockSize]; + size_t dummy; + int error; + + // It's an error to write after closed. + rv = client_ssl_->Write(block, dummy, error); + ASSERT_EQ(rtc::SR_ERROR, rv); + + // But after closed read gives you EOS. + rv = client_ssl_->Read(block, dummy, error); + ASSERT_EQ(rtc::SR_EOS, rv); +} + +// Test a handshake with a bogus peer digest +TEST_P(SSLStreamAdapterTestTLS, TestTLSBogusDigest) { + SetPeerIdentitiesByDigest(false, true); + TestHandshake(false); +} + +TEST_P(SSLStreamAdapterTestTLS, TestTLSDelayedIdentity) { + TestHandshakeWithDelayedIdentity(true); +} + +TEST_P(SSLStreamAdapterTestTLS, TestTLSDelayedIdentityWithBogusDigest) { + TestHandshakeWithDelayedIdentity(false); +} + +// Test that the correct error is returned when SetPeerCertificateDigest is +// called with an unknown algorithm. +TEST_P(SSLStreamAdapterTestTLS, + TestSetPeerCertificateDigestWithUnknownAlgorithm) { + unsigned char server_digest[20]; + size_t server_digest_len; + bool rv; + rtc::SSLPeerCertificateDigestError err; + + rv = server_identity()->certificate().ComputeDigest( + rtc::DIGEST_SHA_1, server_digest, 20, &server_digest_len); + ASSERT_TRUE(rv); + + rv = client_ssl_->SetPeerCertificateDigest("unknown algorithm", server_digest, + server_digest_len, &err); + EXPECT_EQ(rtc::SSLPeerCertificateDigestError::UNKNOWN_ALGORITHM, err); + EXPECT_FALSE(rv); +} + +// Test that the correct error is returned when SetPeerCertificateDigest is +// called with an invalid digest length. +TEST_P(SSLStreamAdapterTestTLS, TestSetPeerCertificateDigestWithInvalidLength) { + unsigned char server_digest[20]; + size_t server_digest_len; + bool rv; + rtc::SSLPeerCertificateDigestError err; + + rv = server_identity()->certificate().ComputeDigest( + rtc::DIGEST_SHA_1, server_digest, 20, &server_digest_len); + ASSERT_TRUE(rv); + + rv = client_ssl_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, server_digest, + server_digest_len - 1, &err); + EXPECT_EQ(rtc::SSLPeerCertificateDigestError::INVALID_LENGTH, err); + EXPECT_FALSE(rv); +} + +// Test moving a bunch of data + +// Basic tests: DTLS +// Test that we can make a handshake work +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) { + TestHandshake(); +} + +// Test that we can make a handshake work if the first packet in +// each direction is lost. This gives us predictable loss +// rather than having to tune random +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacket) { + SetLoseFirstPacket(true); + TestHandshake(); +} + +// Test a handshake with loss and delay +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) { + SetLoseFirstPacket(true); + SetDelay(2000); + SetHandshakeWait(20000); + TestHandshake(); +} + +// Test a handshake with small MTU +// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3910 +TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) { + SetMtu(700); + SetHandshakeWait(20000); + TestHandshake(); +} + +// Test a handshake with total loss and timing out. +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectTimeout) { + SetLoss(100); + TestHandshakeTimeout(); +} + +// Test transfer -- trivial +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransfer) { + TestHandshake(); + TestTransfer(100); +} + +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) { + TestHandshake(); + SetLoss(10); + TestTransfer(100); +} + +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) { + SetDamage(); // Must be called first because first packet + // write happens at end of handshake. + TestHandshake(); + TestTransfer(100); +} + +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSDelayedIdentity) { + TestHandshakeWithDelayedIdentity(true); +} + +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSDelayedIdentityWithBogusDigest) { + TestHandshakeWithDelayedIdentity(false); +} + +// Test DTLS-SRTP with all high ciphers +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) { + std::vector<int> high; + high.push_back(rtc::kSrtpAes128CmSha1_80); + SetDtlsSrtpCryptoSuites(high, true); + SetDtlsSrtpCryptoSuites(high, false); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_EQ(client_cipher, rtc::kSrtpAes128CmSha1_80); +} + +// Test DTLS-SRTP with all low ciphers +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) { + std::vector<int> low; + low.push_back(rtc::kSrtpAes128CmSha1_32); + SetDtlsSrtpCryptoSuites(low, true); + SetDtlsSrtpCryptoSuites(low, false); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_EQ(client_cipher, rtc::kSrtpAes128CmSha1_32); +} + +// Test DTLS-SRTP with a mismatch -- should not converge +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) { + std::vector<int> high; + high.push_back(rtc::kSrtpAes128CmSha1_80); + std::vector<int> low; + low.push_back(rtc::kSrtpAes128CmSha1_32); + SetDtlsSrtpCryptoSuites(high, true); + SetDtlsSrtpCryptoSuites(low, false); + TestHandshake(); + + int client_cipher; + ASSERT_FALSE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_FALSE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); +} + +// Test DTLS-SRTP with each side being mixed -- should select high +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) { + std::vector<int> mixed; + mixed.push_back(rtc::kSrtpAes128CmSha1_80); + mixed.push_back(rtc::kSrtpAes128CmSha1_32); + SetDtlsSrtpCryptoSuites(mixed, true); + SetDtlsSrtpCryptoSuites(mixed, false); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_EQ(client_cipher, rtc::kSrtpAes128CmSha1_80); +} + +// Test DTLS-SRTP with all GCM-128 ciphers. +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) { + std::vector<int> gcm128; + gcm128.push_back(rtc::kSrtpAeadAes128Gcm); + SetDtlsSrtpCryptoSuites(gcm128, true); + SetDtlsSrtpCryptoSuites(gcm128, false); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_EQ(client_cipher, rtc::kSrtpAeadAes128Gcm); +} + +// Test DTLS-SRTP with all GCM-256 ciphers. +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) { + std::vector<int> gcm256; + gcm256.push_back(rtc::kSrtpAeadAes256Gcm); + SetDtlsSrtpCryptoSuites(gcm256, true); + SetDtlsSrtpCryptoSuites(gcm256, false); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_EQ(client_cipher, rtc::kSrtpAeadAes256Gcm); +} + +// Test DTLS-SRTP with mixed GCM-128/-256 ciphers -- should not converge. +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) { + std::vector<int> gcm128; + gcm128.push_back(rtc::kSrtpAeadAes128Gcm); + std::vector<int> gcm256; + gcm256.push_back(rtc::kSrtpAeadAes256Gcm); + SetDtlsSrtpCryptoSuites(gcm128, true); + SetDtlsSrtpCryptoSuites(gcm256, false); + TestHandshake(); + + int client_cipher; + ASSERT_FALSE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_FALSE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); +} + +// Test DTLS-SRTP with both GCM-128/-256 ciphers -- should select GCM-256. +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMixed) { + std::vector<int> gcmBoth; + gcmBoth.push_back(rtc::kSrtpAeadAes256Gcm); + gcmBoth.push_back(rtc::kSrtpAeadAes128Gcm); + SetDtlsSrtpCryptoSuites(gcmBoth, true); + SetDtlsSrtpCryptoSuites(gcmBoth, false); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_EQ(client_cipher, rtc::kSrtpAeadAes256Gcm); +} + +// Test SRTP cipher suite lengths. +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpKeyAndSaltLengths) { + int key_len; + int salt_len; + + ASSERT_FALSE(rtc::GetSrtpKeyAndSaltLengths(rtc::kSrtpInvalidCryptoSuite, + &key_len, &salt_len)); + + ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::kSrtpAes128CmSha1_32, &key_len, + &salt_len)); + ASSERT_EQ(128 / 8, key_len); + ASSERT_EQ(112 / 8, salt_len); + + ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::kSrtpAes128CmSha1_80, &key_len, + &salt_len)); + ASSERT_EQ(128 / 8, key_len); + ASSERT_EQ(112 / 8, salt_len); + + ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::kSrtpAeadAes128Gcm, &key_len, + &salt_len)); + ASSERT_EQ(128 / 8, key_len); + ASSERT_EQ(96 / 8, salt_len); + + ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::kSrtpAeadAes256Gcm, &key_len, + &salt_len)); + ASSERT_EQ(256 / 8, key_len); + ASSERT_EQ(96 / 8, salt_len); +} + +// Test an exporter +TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) { + TestHandshake(); + unsigned char client_out[20]; + unsigned char server_out[20]; + + bool result; + result = ExportKeyingMaterial(kExporterLabel, kExporterContext, + kExporterContextLen, true, true, client_out, + sizeof(client_out)); + ASSERT_TRUE(result); + + result = ExportKeyingMaterial(kExporterLabel, kExporterContext, + kExporterContextLen, true, false, server_out, + sizeof(server_out)); + ASSERT_TRUE(result); + + ASSERT_TRUE(!memcmp(client_out, server_out, sizeof(client_out))); +} + +// Test not yet valid certificates are not rejected. +TEST_P(SSLStreamAdapterTestDTLS, TestCertNotYetValid) { + long one_day = 60 * 60 * 24; + // Make the certificates not valid until one day later. + ResetIdentitiesWithValidity(one_day, one_day); + TestHandshake(); +} + +// Test expired certificates are not rejected. +TEST_P(SSLStreamAdapterTestDTLS, TestCertExpired) { + long one_day = 60 * 60 * 24; + // Make the certificates already expired. + ResetIdentitiesWithValidity(-one_day, -one_day); + TestHandshake(); +} + +// Test data transfer using certs created from strings. +TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestTransfer) { + TestHandshake(); + TestTransfer(100); +} + +// Test getting the remote certificate. +TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) { + // Peer certificates haven't been received yet. + ASSERT_FALSE(GetPeerCertificate(true)); + ASSERT_FALSE(GetPeerCertificate(false)); + + TestHandshake(); + + // The client should have a peer certificate after the handshake. + std::unique_ptr<rtc::SSLCertificate> client_peer_cert = + GetPeerCertificate(true); + ASSERT_TRUE(client_peer_cert); + + // It's not kCERT_PEM. + std::string client_peer_string = client_peer_cert->ToPEMString(); + ASSERT_NE(kCERT_PEM, client_peer_string); + + // The server should have a peer certificate after the handshake. + std::unique_ptr<rtc::SSLCertificate> server_peer_cert = + GetPeerCertificate(false); + ASSERT_TRUE(server_peer_cert); + + // It's kCERT_PEM + ASSERT_EQ(kCERT_PEM, server_peer_cert->ToPEMString()); +} + +// Test getting the used DTLS 1.2 ciphers. +// DTLS 1.2 enabled for client and server -> DTLS 1.2 will be used. +TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Both) { + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher)); + + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_12, GetSslVersion(true)); + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_12, GetSslVersion(false)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_TRUE(rtc::SSLStreamAdapter::IsAcceptableCipher( + server_cipher, ::testing::get<1>(GetParam()).type())); +} + +// Test getting the used DTLS ciphers. +// DTLS 1.2 is max version for client and server. +TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuite) { + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher)); + + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_12, GetSslVersion(true)); + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_12, GetSslVersion(false)); + + ASSERT_EQ(client_cipher, server_cipher); + ASSERT_TRUE(rtc::SSLStreamAdapter::IsAcceptableCipher( + server_cipher, ::testing::get<1>(GetParam()).type())); +} + +// The RSA keysizes here might look strange, why not include the RFC's size +// 2048?. The reason is test case slowness; testing two sizes to exercise +// parametrization is sufficient. +INSTANTIATE_TEST_SUITE_P( + SSLStreamAdapterTestsTLS, + SSLStreamAdapterTestTLS, + Combine(Values(rtc::KeyParams::RSA(1024, 65537), + rtc::KeyParams::RSA(1152, 65537), + rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)), + Values(rtc::KeyParams::RSA(1024, 65537), + rtc::KeyParams::RSA(1152, 65537), + rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)))); +INSTANTIATE_TEST_SUITE_P( + SSLStreamAdapterTestsDTLS, + SSLStreamAdapterTestDTLS, + Combine(Values(rtc::KeyParams::RSA(1024, 65537), + rtc::KeyParams::RSA(1152, 65537), + rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)), + Values(rtc::KeyParams::RSA(1024, 65537), + rtc::KeyParams::RSA(1152, 65537), + rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)))); + +// Tests for enabling / disabling legacy TLS protocols in DTLS. +class SSLStreamAdapterTestDTLSLegacyProtocols + : public SSLStreamAdapterTestDTLSBase { + public: + SSLStreamAdapterTestDTLSLegacyProtocols() + : SSLStreamAdapterTestDTLSBase(rtc::KeyParams::ECDSA(rtc::EC_NIST_P256), + rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)) { + } + + // Do not use the SetUp version from the parent class. + void SetUp() override {} + + // The legacy TLS protocols flag is read when the OpenSSLStreamAdapter is + // initialized, so we set the experiment while creationg client_ssl_ + // and server_ssl_. + + void ConfigureClient(absl::string_view experiment) { + webrtc::test::ScopedFieldTrials trial{std::string(experiment)}; + client_stream_ = + new SSLDummyStreamDTLS(this, "c2s", &client_buffer_, &server_buffer_); + client_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(client_stream_)); + client_ssl_->SignalEvent.connect( + static_cast<SSLStreamAdapterTestBase*>(this), + &SSLStreamAdapterTestBase::OnEvent); + auto client_identity = rtc::SSLIdentity::Create("client", client_key_type_); + client_ssl_->SetIdentity(std::move(client_identity)); + } + + void ConfigureServer(absl::string_view experiment) { + webrtc::test::ScopedFieldTrials trial{std::string(experiment)}; + server_stream_ = + new SSLDummyStreamDTLS(this, "s2c", &server_buffer_, &client_buffer_); + server_ssl_ = + rtc::SSLStreamAdapter::Create(absl::WrapUnique(server_stream_)); + server_ssl_->SignalEvent.connect( + static_cast<SSLStreamAdapterTestBase*>(this), + &SSLStreamAdapterTestBase::OnEvent); + server_ssl_->SetIdentity( + rtc::SSLIdentity::Create("server", server_key_type_)); + } +}; + +// Test getting the used DTLS ciphers. +// DTLS 1.2 enabled for neither client nor server -> DTLS 1.0 will be used. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, TestGetSslCipherSuite) { + ConfigureClient("WebRTC-LegacyTlsProtocols/Enabled/"); + ConfigureServer("WebRTC-LegacyTlsProtocols/Enabled/"); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher)); + + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_10, GetSslVersion(true)); + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_10, GetSslVersion(false)); + + ASSERT_EQ(client_cipher, server_cipher); +} + +// Test getting the used DTLS 1.2 ciphers. +// DTLS 1.2 enabled for client and server -> DTLS 1.2 will be used. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslCipherSuiteDtls12Both) { + ConfigureClient(""); + ConfigureServer(""); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher)); + + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_12, GetSslVersion(true)); + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_12, GetSslVersion(false)); + + ASSERT_EQ(client_cipher, server_cipher); +} + +// DTLS 1.2 enabled for client only -> DTLS 1.0 will be used. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslCipherSuiteDtls12Client) { + ConfigureClient("WebRTC-LegacyTlsProtocols/Enabled/"); + ConfigureServer("WebRTC-LegacyTlsProtocols/Enabled/"); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher)); + + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_10, GetSslVersion(true)); + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_10, GetSslVersion(false)); + + ASSERT_EQ(client_cipher, server_cipher); +} + +// DTLS 1.2 enabled for server only -> DTLS 1.0 will be used. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslCipherSuiteDtls12Server) { + ConfigureClient("WebRTC-LegacyTlsProtocols/Enabled/"); + ConfigureServer("WebRTC-LegacyTlsProtocols/Enabled/"); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(); + + int client_cipher; + ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher)); + int server_cipher; + ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher)); + + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_10, GetSslVersion(true)); + ASSERT_EQ(rtc::SSL_PROTOCOL_DTLS_10, GetSslVersion(false)); + + ASSERT_EQ(client_cipher, server_cipher); +} + +// Client has legacy TLS versions disabled, server has DTLS 1.0 only. +// This is meant to cause a failure. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyDisabledServer10) { + ConfigureClient(""); + ConfigureServer("WebRTC-LegacyTlsProtocols/Enabled/"); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12); + // Handshake should fail. + TestHandshake(false); +} + +// Both client and server have legacy TLS versions disabled and support +// DTLS 1.2. This should work. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyDisabledServer12) { + ConfigureClient(""); + ConfigureServer(""); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12); + TestHandshake(); +} + +// Both client and server have legacy TLS versions enabled and support DTLS 1.0. +// This should work. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyEnabledClient10Server10) { + ConfigureClient("WebRTC-LegacyTlsProtocols/Enabled/"); + ConfigureServer("WebRTC-LegacyTlsProtocols/Enabled/"); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(); +} + +// Legacy protocols are disabled in the client, max TLS version is 1.0 +// This should be a configuration error, and handshake should fail. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyDisabledClient10Server10) { + ConfigureClient(""); + ConfigureServer("WebRTC-LegacyTlsProtocols/Enabled/"); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(false); +} + +// Both client and server have legacy TLS versions enabled and support DTLS 1.0. +// This should work. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyOverrideEnabledClient10Server10) { + rtc::SetAllowLegacyTLSProtocols(true); + ConfigureClient(""); + ConfigureServer(""); + // Remove override. + rtc::SetAllowLegacyTLSProtocols(absl::nullopt); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(); +} + +// Client has legacy TLS disabled and server has legacy TLS enabled via +// override. Handshake for DTLS 1.0 should fail. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyOverrideDisabledClient10EnabledServer10) { + rtc::SetAllowLegacyTLSProtocols(false); + ConfigureClient(""); + rtc::SetAllowLegacyTLSProtocols(true); + ConfigureServer(""); + // Remove override. + rtc::SetAllowLegacyTLSProtocols(absl::nullopt); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(false); +} + +// Client has legacy TLS enabled and server has legacy TLS disabled via +// override. Handshake for DTLS 1.0 should fail. +TEST_F(SSLStreamAdapterTestDTLSLegacyProtocols, + TestGetSslVersionLegacyOverrideEnabledClient10DisabledServer10) { + rtc::SetAllowLegacyTLSProtocols(true); + ConfigureClient(""); + rtc::SetAllowLegacyTLSProtocols(false); + ConfigureServer(""); + // Remove override. + rtc::SetAllowLegacyTLSProtocols(absl::nullopt); + SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); + TestHandshake(false); +} + +// These tests are a no-op under OpenSSL. +#ifdef OPENSSL_IS_BORINGSSL +// TODO(https://bugs.webrtc.org/10261): when removing +// SSLStreamAdapterTestDTLSLegacyProtocols that this class +// inherits from move the code to this class. +class SSLStreamAdapterTestDTLSExtensionPermutation + : public SSLStreamAdapterTestDTLSLegacyProtocols { + public: + SSLStreamAdapterTestDTLSExtensionPermutation() + : SSLStreamAdapterTestDTLSLegacyProtocols() {} +}; + +// Tests for enabling the (D)TLS extension permutation which randomizes the +// order of extensions in the client hello. +TEST_F(SSLStreamAdapterTestDTLSExtensionPermutation, + ClientDefaultServerDefault) { + ConfigureClient(""); + ConfigureServer(""); + TestHandshake(); +} + +TEST_F(SSLStreamAdapterTestDTLSExtensionPermutation, + ClientDefaultServerPermute) { + ConfigureClient(""); + ConfigureServer("WebRTC-PermuteTlsClientHello/Enabled/"); + TestHandshake(); +} + +TEST_F(SSLStreamAdapterTestDTLSExtensionPermutation, + ClientPermuteServerDefault) { + ConfigureClient("WebRTC-PermuteTlsClientHello/Enabled/"); + ConfigureServer(""); + TestHandshake(); +} + +TEST_F(SSLStreamAdapterTestDTLSExtensionPermutation, + ClientPermuteServerPermute) { + ConfigureClient("WebRTC-PermuteTlsClientHello/Enabled/"); + ConfigureServer("WebRTC-PermuteTlsClientHello/Enabled/"); + TestHandshake(); +} +#endif // OPENSSL_IS_BORINGSSL |