zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL.

Browse Source 
These defines don't work any more, so they only cause confusion:

FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL

BUG=webrtc:7025

Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
This commit is contained in:
deadbeef
2017-01-23 19:39:57 -08:00
committed by Commit bot
parent 4c78702d12
commit 1b54a5f018
24 changed files with 14 additions and 447 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/BUILD.gn
View File

@ -30,7 +30,6 @@ config("common_inherited_config") {
defines = [
# TODO(kjellander): Cleanup unused ones and move defines closer to
# the source when webrtc:4256 is completed.
"FEATURE_ENABLE_SSL",
"FEATURE_ENABLE_VOICEMAIL",
"EXPAT_RELATIVE_PATH",
"GTEST_RELATIVE_PATH",
@ -132,12 +131,10 @@ config("common_config") {
# targets, there's no point including the defines in that config here.
# TODO(kjellander): Cleanup unused ones and move defines closer to the
# source when webrtc:4256 is completed.
"HAVE_OPENSSL_SSL_H",
"HAVE_SRTP",
"HAVE_WEBRTC_VIDEO",
"HAVE_WEBRTC_VOICE",
"LOGGING_INSIDE_WEBRTC",
"SSL_USE_OPENSSL",
"USE_WEBRTC_DEV_BRANCH",
]
} else {

23
webrtc/base/BUILD.gn
View File

@ -35,21 +35,10 @@ config("rtc_base_approved_all_dependent_config") {
}
}
config("rtc_base_config") {
defines = [ "FEATURE_ENABLE_SSL" ]
}
config("rtc_base_chromium_config") {
defines = [ "NO_MAIN_THREAD_WRAPPING" ]
}
config("openssl_config") {
defines = [
"SSL_USE_OPENSSL",
"HAVE_OPENSSL_SSL_H",
]
}
config("rtc_base_all_dependent_config") {
if (is_ios) {
libs = [
@ -370,16 +359,7 @@ rtc_static_library("rtc_base") {
public_deps = [
":rtc_base_approved",
]
configs += [
":openssl_config",
":rtc_base_config",
]
public_configs = [
":openssl_config",
":rtc_base_config",
]
public_configs = []
all_dependent_configs = [ ":rtc_base_all_dependent_config" ]
@ -537,7 +517,6 @@ rtc_static_library("rtc_base") {
"proxyserver.h",
"rollingaccumulator.h",
"scopedptrcollection.h",
"sslconfig.h",
"sslroots.h",
"testbase64.h",
"testclient.cc",

85
webrtc/base/helpers.cc
View File

@ -13,18 +13,7 @@
#include <limits>
#include <memory>
#if defined(FEATURE_ENABLE_SSL)
#include "webrtc/base/sslconfig.h"
#if defined(SSL_USE_OPENSSL)
#include <openssl/rand.h>
#else
#if defined(WEBRTC_WIN)
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <ntsecapi.h>
#endif // WEBRTC_WIN
#endif // else
#endif // FEATURE_ENABLED_SSL
#include "webrtc/base/base64.h"
#include "webrtc/base/basictypes.h"
@ -45,7 +34,6 @@ class RandomGenerator {
virtual bool Generate(void* buf, size_t len) = 0;
};
#if defined(SSL_USE_OPENSSL)
// The OpenSSL RNG.
class SecureRandomGenerator : public RandomGenerator {
public:
@ -57,79 +45,6 @@ class SecureRandomGenerator : public RandomGenerator {
}
};
#else
#if defined(WEBRTC_WIN)
class SecureRandomGenerator : public RandomGenerator {
public:
SecureRandomGenerator() : advapi32_(NULL), rtl_gen_random_(NULL) {}
~SecureRandomGenerator() {
FreeLibrary(advapi32_);
}
virtual bool Init(const void* seed, size_t seed_len) {
// We don't do any additional seeding on Win32, we just use the CryptoAPI
// RNG (which is exposed as a hidden function off of ADVAPI32 so that we
// don't need to drag in all of CryptoAPI)
if (rtl_gen_random_) {
return true;
}
advapi32_ = LoadLibrary(L"advapi32.dll");
if (!advapi32_) {
return false;
}
rtl_gen_random_ = reinterpret_cast<RtlGenRandomProc>(
GetProcAddress(advapi32_, "SystemFunction036"));
if (!rtl_gen_random_) {
FreeLibrary(advapi32_);
return false;
}
return true;
}
virtual bool Generate(void* buf, size_t len) {
if (!rtl_gen_random_ && !Init(NULL, 0)) {
return false;
}
return (rtl_gen_random_(buf, static_cast<int>(len)) != FALSE);
}
private:
typedef BOOL (WINAPI *RtlGenRandomProc)(PVOID, ULONG);
HINSTANCE advapi32_;
RtlGenRandomProc rtl_gen_random_;
};
#elif !defined(FEATURE_ENABLE_SSL)
// No SSL implementation -- use rand()
class SecureRandomGenerator : public RandomGenerator {
public:
virtual bool Init(const void* seed, size_t len) {
if (len >= 4) {
srand(*reinterpret_cast<const int*>(seed));
} else {
srand(*reinterpret_cast<const char*>(seed));
}
return true;
}
virtual bool Generate(void* buf, size_t len) {
char* bytes = reinterpret_cast<char*>(buf);
for (size_t i = 0; i < len; ++i) {
bytes[i] = static_cast<char>(rand());
}
return true;
}
};
#else
#error No SSL implementation has been selected!
#endif // WEBRTC_WIN
#endif
// A test random generator, for predictable output.
class TestRandomGenerator : public RandomGenerator {
public:

16
webrtc/base/messagedigest.cc
View File

@ -15,13 +15,7 @@
#include <string.h>
#include "webrtc/base/basictypes.h"
#include "webrtc/base/sslconfig.h"
#if SSL_USE_OPENSSL
#include "webrtc/base/openssldigest.h"
#else
#include "webrtc/base/md5digest.h"
#include "webrtc/base/sha1digest.h"
#endif
#include "webrtc/base/stringencode.h"
namespace rtc {
@ -37,22 +31,12 @@ const char DIGEST_SHA_512[] = "sha-512";
static const size_t kBlockSize = 64; // valid for SHA-256 and down
MessageDigest* MessageDigestFactory::Create(const std::string& alg) {
#if SSL_USE_OPENSSL
MessageDigest* digest = new OpenSSLDigest(alg);
if (digest->Size() == 0) { // invalid algorithm
delete digest;
digest = NULL;
}
return digest;
#else
MessageDigest* digest = NULL;
if (alg == DIGEST_MD5) {
digest = new Md5Digest();
} else if (alg == DIGEST_SHA_1) {
digest = new Sha1Digest();
}
return digest;
#endif
}
bool IsFips180DigestAlgorithm(const std::string& alg) {

4
webrtc/base/openssladapter.cc
View File

@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#if HAVE_OPENSSL_SSL_H
#include "webrtc/base/openssladapter.h"
#if defined(WEBRTC_POSIX)
@ -965,5 +963,3 @@ OpenSSLAdapter::SetupSSLContext() {
}
} // namespace rtc
#endif // HAVE_OPENSSL_SSL_H

4
webrtc/base/openssldigest.cc
View File

@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#if HAVE_OPENSSL_SSL_H
#include "webrtc/base/openssldigest.h"
#include "webrtc/base/checks.h"
@ -118,5 +116,3 @@ bool OpenSSLDigest::GetDigestSize(const std::string& algorithm,
}
} // namespace rtc
#endif // HAVE_OPENSSL_SSL_H

4
webrtc/base/opensslidentity.cc
View File

@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#if HAVE_OPENSSL_SSL_H
#include "webrtc/base/opensslidentity.h"
#include <memory>
@ -576,5 +574,3 @@ bool OpenSSLIdentity::operator!=(const OpenSSLIdentity& other) const {
}
} // namespace rtc
#endif // HAVE_OPENSSL_SSL_H

44
webrtc/base/opensslstreamadapter.cc
View File

@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#if HAVE_OPENSSL_SSL_H
#include "webrtc/base/opensslstreamadapter.h"
#include <openssl/bio.h>
@ -45,11 +43,10 @@ namespace {
namespace rtc {
#if (OPENSSL_VERSION_NUMBER >= 0x10001000L)
#define HAVE_DTLS_SRTP
#if (OPENSSL_VERSION_NUMBER < 0x10001000L)
#error "webrtc requires at least OpenSSL version 1.0.1, to support DTLS-SRTP"
#endif
#ifdef HAVE_DTLS_SRTP
// SRTP cipher suite table. |internal_name| is used to construct a
// colon-separated profile strings which is needed by
// SSL_CTX_set_tlsext_use_srtp().
@ -65,7 +62,6 @@ static SrtpCipherMapEntry SrtpCipherMap[] = {
{"SRTP_AEAD_AES_128_GCM", SRTP_AEAD_AES_128_GCM},
{"SRTP_AEAD_AES_256_GCM", SRTP_AEAD_AES_256_GCM},
{nullptr, 0}};
#endif
#ifdef OPENSSL_IS_BORINGSSL
// Not used in production code. Actual time should be relative to Jan 1, 1970.
@ -432,7 +428,6 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
bool use_context,
uint8_t* result,
size_t result_len) {
#ifdef HAVE_DTLS_SRTP
int i;
i = SSL_export_keying_material(ssl_, result, result_len, label.c_str(),
@ -443,14 +438,10 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
return false;
return true;
#else
return false;
#endif
}
bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
const std::vector<int>& ciphers) {
#ifdef HAVE_DTLS_SRTP
std::string internal_ciphers;
if (state_ != SSL_NONE)
@ -481,13 +472,9 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
srtp_ciphers_ = internal_ciphers;
return true;
#else
return false;
#endif
}
bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
#ifdef HAVE_DTLS_SRTP
RTC_DCHECK(state_ == SSL_CONNECTED);
if (state_ != SSL_CONNECTED)
return false;
@ -501,9 +488,6 @@ bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
*crypto_suite = srtp_profile->id;
RTC_DCHECK(!SrtpCryptoSuiteToName(*crypto_suite).empty());
return true;
#else
return false;
#endif
}
bool OpenSSLStreamAdapter::IsTlsConnected() {
@ -1096,14 +1080,12 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
SSL_CTX_set_cipher_list(ctx,
"DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK");
#ifdef HAVE_DTLS_SRTP
if (!srtp_ciphers_.empty()) {
if (SSL_CTX_set_tlsext_use_srtp(ctx, srtp_ciphers_.c_str())) {
SSL_CTX_free(ctx);
return NULL;
}
}
#endif
return ctx;
}
@ -1169,26 +1151,6 @@ int OpenSSLStreamAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
return stream->VerifyPeerCertificate();
}
bool OpenSSLStreamAdapter::HaveDtls() {
return true;
}
bool OpenSSLStreamAdapter::HaveDtlsSrtp() {
#ifdef HAVE_DTLS_SRTP
return true;
#else
return false;
#endif
}
bool OpenSSLStreamAdapter::HaveExporter() {
#ifdef HAVE_DTLS_SRTP
return true;
#else
return false;
#endif
}
bool OpenSSLStreamAdapter::IsBoringSsl() {
#ifdef OPENSSL_IS_BORINGSSL
return true;
@ -1273,5 +1235,3 @@ void OpenSSLStreamAdapter::enable_time_callback_for_testing() {
}
} // namespace rtc
#endif // HAVE_OPENSSL_SSL_H

5
webrtc/base/opensslstreamadapter.h
View File

@ -109,10 +109,7 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
bool IsTlsConnected() override;
// Capabilities interfaces
static bool HaveDtls();
static bool HaveDtlsSrtp();
static bool HaveExporter();
// Capabilities interfaces.
static bool IsBoringSsl();
static bool IsAcceptableCipher(int cipher, KeyType key_type);

31
webrtc/base/ssladapter.cc
View File

@ -10,13 +10,7 @@
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslconfig.h"
#if SSL_USE_OPENSSL
#include "openssladapter.h"
#endif
#include "webrtc/base/openssladapter.h"
///////////////////////////////////////////////////////////////////////////////
@ -24,18 +18,11 @@ namespace rtc {
SSLAdapter*
SSLAdapter::Create(AsyncSocket* socket) {
#if SSL_USE_OPENSSL
return new OpenSSLAdapter(socket);
#else // !SSL_USE_OPENSSL
delete socket;
return NULL;
#endif // SSL_USE_OPENSSL
}
///////////////////////////////////////////////////////////////////////////////
#if SSL_USE_OPENSSL
bool InitializeSSL(VerificationCallback callback) {
return OpenSSLAdapter::InitializeSSL(callback);
}
@ -48,22 +35,6 @@ bool CleanupSSL() {
return OpenSSLAdapter::CleanupSSL();
}
#else // !SSL_USE_OPENSSL
bool InitializeSSL(VerificationCallback callback) {
return true;
}
bool InitializeSSLThread() {
return true;
}
bool CleanupSSL() {
return true;
}
#endif // SSL_USE_OPENSSL
///////////////////////////////////////////////////////////////////////////////
} // namespace rtc

4
webrtc/base/ssladapter_unittest.cc
View File

@ -370,8 +370,6 @@ class SSLAdapterTestDTLS_ECDSA : public SSLAdapterTestBase {
: SSLAdapterTestBase(rtc::SSL_MODE_DTLS, rtc::KeyParams::ECDSA()) {}
};
#if SSL_USE_OPENSSL
// Basic tests: TLS
// Test that handshake works, using RSA
@ -419,5 +417,3 @@ TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSTransfer) {
TestHandshake(true);
TestTransfer("Hello, world!");
}
#endif // SSL_USE_OPENSSL

30
webrtc/base/sslconfig.h
View File

@ -1,30 +0,0 @@
/*
* Copyright 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.
*/
#ifndef WEBRTC_BASE_SSLCONFIG_H_
#define WEBRTC_BASE_SSLCONFIG_H_
// If no preference has been indicated, default to SChannel on Windows and
// OpenSSL everywhere else, if it is available.
#if !defined(SSL_USE_SCHANNEL) && !defined(SSL_USE_OPENSSL)
#if defined(WEBRTC_WIN)
#define SSL_USE_SCHANNEL 1
#else // defined(WEBRTC_WIN)
#if defined(HAVE_OPENSSL_SSL_H)
#define SSL_USE_OPENSSL 1
#endif
#endif // !defined(WEBRTC_WIN)
#endif
#endif // WEBRTC_BASE_SSLCONFIG_H_

16
webrtc/base/sslidentity.cc
View File

@ -17,14 +17,8 @@
#include "webrtc/base/base64.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/sslconfig.h"
#include "webrtc/base/sslfingerprint.h"
#if SSL_USE_OPENSSL
#include "webrtc/base/opensslidentity.h"
#endif // SSL_USE_OPENSSL
#include "webrtc/base/sslfingerprint.h"
namespace rtc {
@ -213,8 +207,6 @@ SSLCertChain::~SSLCertChain() {
std::for_each(certs_.begin(), certs_.end(), DeleteCert);
}
#if SSL_USE_OPENSSL
// static
SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
return OpenSSLCertificate::FromPEMString(pem_string);
@ -260,12 +252,6 @@ bool operator!=(const SSLIdentity& a, const SSLIdentity& b) {
return !(a == b);
}
#else // !SSL_USE_OPENSSL
#error "No SSL implementation"
#endif // SSL_USE_OPENSSL
// Read |n| bytes from ASN1 number string at *|pp| and return the numeric value.
// Update *|pp| and *|np| to reflect number of read bytes.
static inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) {

20
webrtc/base/sslstreamadapter.cc
View File

@ -9,14 +9,9 @@
*/
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/sslconfig.h"
#if SSL_USE_OPENSSL
#include "webrtc/base/opensslstreamadapter.h"
#endif // SSL_USE_OPENSSL
///////////////////////////////////////////////////////////////////////////////
namespace rtc {
@ -101,11 +96,7 @@ CryptoOptions CryptoOptions::NoGcm() {
}
SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
#if SSL_USE_OPENSSL
return new OpenSSLStreamAdapter(stream);
#else // !SSL_USE_OPENSSL
return NULL;
#endif // SSL_USE_OPENSSL
}
SSLStreamAdapter::SSLStreamAdapter(StreamInterface* stream)
@ -137,16 +128,6 @@ bool SSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
return false;
}
#if SSL_USE_OPENSSL
bool SSLStreamAdapter::HaveDtls() {
return OpenSSLStreamAdapter::HaveDtls();
}
bool SSLStreamAdapter::HaveDtlsSrtp() {
return OpenSSLStreamAdapter::HaveDtlsSrtp();
}
bool SSLStreamAdapter::HaveExporter() {
return OpenSSLStreamAdapter::HaveExporter();
}
bool SSLStreamAdapter::IsBoringSsl() {
return OpenSSLStreamAdapter::IsBoringSsl();
}
@ -163,7 +144,6 @@ std::string SSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
void SSLStreamAdapter::enable_time_callback_for_testing() {
OpenSSLStreamAdapter::enable_time_callback_for_testing();
}
#endif // SSL_USE_OPENSSL
///////////////////////////////////////////////////////////////////////////////

7
webrtc/base/sslstreamadapter.h
View File

@ -228,10 +228,9 @@ class SSLStreamAdapter : public StreamAdapterInterface {
// SS_OPENING but IsTlsConnected should return true.
virtual bool IsTlsConnected() = 0;
// Capabilities testing
static bool HaveDtls();
static bool HaveDtlsSrtp();
static bool HaveExporter();
// Capabilities testing.
// Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now
// that's assumed.
static bool IsBoringSsl();
// Returns true iff the supplied cipher is deemed to be strong.

32
webrtc/base/sslstreamadapter_unittest.cc
View File

@ -19,7 +19,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslconfig.h"
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stream.h"
@ -65,12 +64,6 @@ static const char kCERT_PEM[] =
"UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
"-----END CERTIFICATE-----\n";
#define MAYBE_SKIP_TEST(feature) \
if (!(rtc::SSLStreamAdapter::feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
class SSLStreamAdapterTestBase;
class SSLDummyStreamBase : public rtc::StreamInterface,
@ -963,7 +956,6 @@ TEST_P(SSLStreamAdapterTestTLS, TestSetPeerCertificateDigestWithInvalidLength) {
// Basic tests: DTLS
// Test that we can make a handshake work
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
};
@ -971,14 +963,12 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) {
// each direction is lost. This gives us predictable loss
// rather than having to tune random
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacket) {
MAYBE_SKIP_TEST(HaveDtls);
SetLoseFirstPacket(true);
TestHandshake();
};
// Test a handshake with loss and delay
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) {
MAYBE_SKIP_TEST(HaveDtls);
SetLoseFirstPacket(true);
SetDelay(2000);
SetHandshakeWait(20000);
@ -988,7 +978,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) {
// Test a handshake with small MTU
// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3910
TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) {
MAYBE_SKIP_TEST(HaveDtls);
SetMtu(700);
SetHandshakeWait(20000);
TestHandshake();
@ -996,20 +985,17 @@ TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) {
// Test transfer -- trivial
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransfer) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
TestTransfer(100);
};
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
SetLoss(10);
TestTransfer(100);
};
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) {
MAYBE_SKIP_TEST(HaveDtls);
SetDamage(); // Must be called first because first packet
// write happens at end of handshake.
TestHandshake();
@ -1026,7 +1012,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSDelayedIdentityWithBogusDigest) {
// Test DTLS-SRTP with all high ciphers
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> high;
high.push_back(rtc::SRTP_AES128_CM_SHA1_80);
SetDtlsSrtpCryptoSuites(high, true);
@ -1044,7 +1029,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) {
// Test DTLS-SRTP with all low ciphers
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> low;
low.push_back(rtc::SRTP_AES128_CM_SHA1_32);
SetDtlsSrtpCryptoSuites(low, true);
@ -1062,7 +1046,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) {
// Test DTLS-SRTP with a mismatch -- should not converge
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> high;
high.push_back(rtc::SRTP_AES128_CM_SHA1_80);
std::vector<int> low;
@ -1079,7 +1062,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) {
// Test DTLS-SRTP with each side being mixed -- should select high
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> mixed;
mixed.push_back(rtc::SRTP_AES128_CM_SHA1_80);
mixed.push_back(rtc::SRTP_AES128_CM_SHA1_32);
@ -1098,7 +1080,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) {
// Test DTLS-SRTP with all GCM-128 ciphers.
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> gcm128;
gcm128.push_back(rtc::SRTP_AEAD_AES_128_GCM);
SetDtlsSrtpCryptoSuites(gcm128, true);
@ -1116,7 +1097,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) {
// Test DTLS-SRTP with all GCM-256 ciphers.
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> gcm256;
gcm256.push_back(rtc::SRTP_AEAD_AES_256_GCM);
SetDtlsSrtpCryptoSuites(gcm256, true);
@ -1134,7 +1114,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) {
// Test DTLS-SRTP with mixed GCM-128/-256 ciphers -- should not converge.
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> gcm128;
gcm128.push_back(rtc::SRTP_AEAD_AES_128_GCM);
std::vector<int> gcm256;
@ -1151,7 +1130,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) {
// Test DTLS-SRTP with both GCM-128/-256 ciphers -- should select GCM-256.
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMixed) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
std::vector<int> gcmBoth;
gcmBoth.push_back(rtc::SRTP_AEAD_AES_256_GCM);
gcmBoth.push_back(rtc::SRTP_AEAD_AES_128_GCM);
@ -1199,7 +1177,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpKeyAndSaltLengths) {
// Test an exporter
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) {
MAYBE_SKIP_TEST(HaveExporter);
TestHandshake();
unsigned char client_out[20];
unsigned char server_out[20];
@ -1222,7 +1199,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) {
// Test not yet valid certificates are not rejected.
TEST_P(SSLStreamAdapterTestDTLS, TestCertNotYetValid) {
MAYBE_SKIP_TEST(HaveDtls);
long one_day = 60 * 60 * 24;
// Make the certificates not valid until one day later.
ResetIdentitiesWithValidity(one_day, one_day);
@ -1231,7 +1207,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestCertNotYetValid) {
// Test expired certificates are not rejected.
TEST_P(SSLStreamAdapterTestDTLS, TestCertExpired) {
MAYBE_SKIP_TEST(HaveDtls);
long one_day = 60 * 60 * 24;
// Make the certificates already expired.
ResetIdentitiesWithValidity(-one_day, -one_day);
@ -1240,15 +1215,12 @@ TEST_P(SSLStreamAdapterTestDTLS, TestCertExpired) {
// Test data transfer using certs created from strings.
TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestTransfer) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
TestTransfer(100);
}
// Test getting the remote certificate.
TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
MAYBE_SKIP_TEST(HaveDtls);
// Peer certificates haven't been received yet.
ASSERT_FALSE(GetPeerCertificate(true));
ASSERT_FALSE(GetPeerCertificate(false));
@ -1282,7 +1254,6 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
// Test getting the used DTLS ciphers.
// DTLS 1.2 enabled for neither client nor server -> DTLS 1.0 will be used.
TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuite) {
MAYBE_SKIP_TEST(HaveDtls);
SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10);
TestHandshake();
@ -1302,7 +1273,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuite) {
// 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) {
MAYBE_SKIP_TEST(HaveDtls);
SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12);
TestHandshake();
@ -1321,7 +1291,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Both) {
// DTLS 1.2 enabled for client only -> DTLS 1.0 will be used.
TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Client) {
MAYBE_SKIP_TEST(HaveDtls);
SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12);
TestHandshake();
@ -1340,7 +1309,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Client) {
// DTLS 1.2 enabled for server only -> DTLS 1.0 will be used.
TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Server) {
MAYBE_SKIP_TEST(HaveDtls);
SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10);
TestHandshake();

2
webrtc/p2p/BUILD.gn
View File

@ -79,7 +79,7 @@ rtc_static_library("rtc_p2p") {
"client/socketmonitor.h",
]
defines = [ "FEATURE_ENABLE_SSL" ]
defines = []
deps = [
"../base:rtc_base",

24
webrtc/p2p/base/dtlstransportchannel_unittest.cc
View File

@ -693,7 +693,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferSrtpTwoChannels) {
// Connect with DTLS, and transfer some data.
TEST_F(DtlsTransportChannelTest, TestTransferDtls) {
MAYBE_SKIP_TEST(HaveDtls);
PrepareDtls(true, true, rtc::KT_DEFAULT);
ASSERT_TRUE(Connect());
TestTransfer(0, 1000, 100, false);
@ -701,7 +700,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtls) {
// Create two channels with DTLS, and transfer some data.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsTwoChannels) {
MAYBE_SKIP_TEST(HaveDtls);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
ASSERT_TRUE(Connect());
@ -725,7 +723,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsNotOffered) {
// Create two channels with DTLS 1.0 and check ciphers.
TEST_F(DtlsTransportChannelTest, TestDtls12None) {
MAYBE_SKIP_TEST(HaveDtls);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10);
@ -734,7 +731,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12None) {
// Create two channels with DTLS 1.2 and check ciphers.
TEST_F(DtlsTransportChannelTest, TestDtls12Both) {
MAYBE_SKIP_TEST(HaveDtls);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12);
@ -743,7 +739,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12Both) {
// Create two channels with DTLS 1.0 / DTLS 1.2 and check ciphers.
TEST_F(DtlsTransportChannelTest, TestDtls12Client1) {
MAYBE_SKIP_TEST(HaveDtls);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10);
@ -752,7 +747,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12Client1) {
// Create two channels with DTLS 1.2 / DTLS 1.0 and check ciphers.
TEST_F(DtlsTransportChannelTest, TestDtls12Client2) {
MAYBE_SKIP_TEST(HaveDtls);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12);
@ -761,7 +755,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12Client2) {
// Connect with DTLS, negotiate DTLS-SRTP, and transfer SRTP using bypass.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
ASSERT_TRUE(Connect());
@ -771,7 +764,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) {
// Connect with DTLS-SRTP, transfer an invalid SRTP packet, and expects -1
// returned.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsInvalidSrtpPacket) {
MAYBE_SKIP_TEST(HaveDtls);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
ASSERT_TRUE(Connect());
@ -781,7 +773,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsInvalidSrtpPacket) {
// Connect with DTLS. A does DTLS-SRTP but B does not.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpRejected) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, false);
ASSERT_TRUE(Connect());
@ -789,7 +780,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpRejected) {
// Connect with DTLS. B does DTLS-SRTP but A does not.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpNotOffered) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(false, true);
ASSERT_TRUE(Connect());
@ -797,7 +787,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpNotOffered) {
// Create two channels with DTLS, negotiate DTLS-SRTP, and transfer bypass SRTP.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpTwoChannels) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -808,7 +797,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpTwoChannels) {
// Create a single channel with DTLS, and send normal data and SRTP data on it.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
ASSERT_TRUE(Connect());
@ -818,7 +806,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) {
// Testing when the remote is passive.
TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -831,7 +818,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) {
// Testing with the legacy DTLS client which doesn't use setup attribute.
// In this case legacy is the answerer.
TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
PrepareDtls(true, true, rtc::KT_DEFAULT);
NegotiateWithLegacy();
rtc::SSLRole channel1_role;
@ -845,7 +831,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) {
// Testing re offer/answer after the session is estbalished. Roles will be
// kept same as of the previous negotiation.
TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -862,7 +847,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) {
}
TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -880,7 +864,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) {
// Test that any change in role after the intial setup will result in failure.
TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -896,7 +879,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) {
// Test that using different setup attributes which results in similar ssl
// role as the initial negotiation will result in success.
TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -912,7 +894,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) {
// Test that re-negotiation can be started before the clients become connected
// in the first negotiation.
TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
SetChannelCount(2);
PrepareDtls(true, true, rtc::KT_DEFAULT);
PrepareDtlsSrtp(true, true);
@ -932,7 +913,6 @@ TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
// Test Certificates state after negotiation but before connection.
TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) {
MAYBE_SKIP_TEST(HaveDtls);
PrepareDtls(true, true, rtc::KT_DEFAULT);
Negotiate();
@ -953,7 +933,6 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) {
// Test Certificates state after connection.
TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
MAYBE_SKIP_TEST(HaveDtls);
PrepareDtls(true, true, rtc::KT_DEFAULT);
ASSERT_TRUE(Connect());
@ -984,7 +963,6 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
// 60 seconds. The timer defaults to 1 second, but for WebRTC we should be
// initializing it to 50ms.
TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) {
MAYBE_SKIP_TEST(HaveDtls);
// We can only change the retransmission schedule with a recently-added
// BoringSSL API. Skip the test if not built with BoringSSL.
MAYBE_SKIP_TEST(IsBoringSsl);
@ -1025,7 +1003,6 @@ TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) {
// Test that a DTLS connection can be made even if the underlying transport
// is connected before DTLS fingerprints/roles have been negotiated.
TEST_F(DtlsTransportChannelTest, TestConnectBeforeNegotiate) {
MAYBE_SKIP_TEST(HaveDtls);
PrepareDtls(true, true, rtc::KT_DEFAULT);
ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
cricket::CONNECTIONROLE_ACTIVE,
@ -1158,7 +1135,6 @@ class DtlsEventOrderingTest
};
TEST_P(DtlsEventOrderingTest, TestEventOrdering) {
MAYBE_SKIP_TEST(HaveDtls);
TestEventOrdering(::testing::get<0>(GetParam()),
::testing::get<1>(GetParam()));
}

24
webrtc/pc/channel_unittest.cc
View File

@ -25,12 +25,6 @@
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/channel.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(rtc::SSLStreamAdapter::feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using cricket::CA_OFFER;
using cricket::CA_PRANSWER;
using cricket::CA_ANSWER;
@ -2243,32 +2237,26 @@ TEST_F(VoiceChannelSingleThreadTest, SendSrtcpMux) {
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, 0);
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, DTLS);
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpGcmBoth) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS | GCM_CIPHER);
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpGcmOne) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS);
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpGcmTwo) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, DTLS | GCM_CIPHER);
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
}
@ -2576,32 +2564,26 @@ TEST_F(VoiceChannelDoubleThreadTest, SendSrtcpMux) {
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, 0);
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, DTLS);
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpGcmBoth) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS | GCM_CIPHER);
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpGcmOne) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS);
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpGcmTwo) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, DTLS | GCM_CIPHER);
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
}
@ -2901,17 +2883,14 @@ TEST_F(VideoChannelSingleThreadTest, SendSrtpToRtp) {
}
TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, 0);
}
TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, DTLS);
}
TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
}
@ -3133,17 +3112,14 @@ TEST_F(VideoChannelDoubleThreadTest, SendSrtpToRtp) {
}
TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, 0);
}
TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS, DTLS);
}
TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
}

22
webrtc/pc/peerconnection_unittest.cc
View File

@ -46,12 +46,6 @@
#include "webrtc/pc/test/fakevideotrackrenderer.h"
#include "webrtc/pc/test/mockpeerconnectionobservers.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using cricket::ContentInfo;
using cricket::FakeWebRtcVideoDecoder;
using cricket::FakeWebRtcVideoDecoderFactory;
@ -223,8 +217,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeRTCCertificateGenerator() : nullptr);
new FakeRTCCertificateGenerator());
return CreateClientWithDtlsIdentityStore(id, constraints, options, config,
std::move(cert_generator), true,
@ -237,8 +230,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeRTCCertificateGenerator() : nullptr);
new FakeRTCCertificateGenerator());
return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr,
std::move(cert_generator), false,
@ -1472,7 +1464,6 @@ class P2PTestConductor : public testing::Test {
}
void SetupAndVerifyDtlsCall() {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@ -1497,8 +1488,7 @@ class P2PTestConductor : public testing::Test {
rtc_config.set_cpu_adaptation(false);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeRTCCertificateGenerator() : nullptr);
new FakeRTCCertificateGenerator());
cert_generator->use_alternate_key();
// Make sure the new client is using a different certificate.
@ -1694,7 +1684,6 @@ TEST_F(P2PTestConductor, OneWayMediaCallWithoutConstraints) {
// This test sets up a audio call initially and then upgrades to audio/video,
// using DTLS.
TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@ -1708,7 +1697,6 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
// This test sets up a call transfer to a new caller with a different DTLS
// fingerprint.
TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetupAndVerifyDtlsCall();
// Keeping the original peer around which will still send packets to the
@ -1727,7 +1715,6 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
// bundle is in effect in the restart, the channel can successfully reset its
// DTLS-SRTP context.
TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@ -1746,7 +1733,6 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
// This test sets up a call transfer to a new callee with a different DTLS
// fingerprint.
TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetupAndVerifyDtlsCall();
// Keeping the original peer around which will still send packets to the
@ -1780,7 +1766,6 @@ TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) {
// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
// negotiated and used for transport.
TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@ -2257,7 +2242,6 @@ TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
// negotiation is completed without error.
#ifdef HAVE_SCTP
TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.SetMandatory(
MediaConstraintsInterface::kEnableDtlsSrtp, true);

18
webrtc/pc/peerconnectionendtoend_unittest.cc
View File

@ -24,12 +24,6 @@
// Notice that mockpeerconnectionobservers.h must be included after the above!
#include "webrtc/pc/test/mockpeerconnectionobservers.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
@ -198,8 +192,6 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
// Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
@ -224,8 +216,6 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
// Verifies that a DataChannel created after the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
@ -257,8 +247,6 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
@ -286,8 +274,6 @@ TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
// there are multiple DataChannels.
TEST_F(PeerConnectionEndToEndTest,
MessageTransferBetweenTwoPairsOfDataChannels) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
@ -409,8 +395,6 @@ TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
TEST_F(PeerConnectionEndToEndTest,
DISABLED_DataChannelFromOpenWorksAfterClose) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
@ -437,8 +421,6 @@ TEST_F(PeerConnectionEndToEndTest,
// reference count), no memory access violation will occur.
// See: https://code.google.com/p/chromium/issues/detail?id=565048
TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;

7
webrtc/pc/peerconnectioninterface_unittest.cc
View File

@ -318,12 +318,6 @@ static const char kDtlsSdesFallbackSdp[] =
"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
"dummy_session_params\r\n";
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using ::testing::Exactly;
using cricket::StreamParams;
using webrtc::AudioSourceInterface;
@ -2069,7 +2063,6 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);

3
webrtc/pc/test/peerconnectiontestwrapper.cc
View File

@ -76,8 +76,7 @@ bool PeerConnectionTestWrapper::CreatePc(
}
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeRTCCertificateGenerator()
: nullptr);
new FakeRTCCertificateGenerator());
peer_connection_ = peer_connection_factory_->CreatePeerConnection(
config, constraints, std::move(port_allocator), std::move(cert_generator),
this);

33
webrtc/pc/webrtcsession_unittest.cc
View File

@ -50,12 +50,6 @@
#include "webrtc/pc/webrtcsession.h"
#include "webrtc/pc/webrtcsessiondescriptionfactory.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using cricket::FakeVoiceMediaChannel;
using cricket::TransportInfo;
using rtc::SocketAddress;
@ -1850,7 +1844,6 @@ TEST_F(WebRtcSessionTest, TestSetRemoteNonSdesAnswerWhenSdesOn) {
// Test that we accept an offer with a DTLS fingerprint when DTLS is on
// and that we return an answer with a DTLS fingerprint.
TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SendAudioVideoStream1();
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@ -1879,7 +1872,6 @@ TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) {
// Test that we set a local offer with a DTLS fingerprint when DTLS is on
// and then we accept a remote answer with a DTLS fingerprint successfully.
TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SendAudioVideoStream1();
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@ -1909,7 +1901,6 @@ TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) {
// Test that if we support DTLS and the other side didn't offer a fingerprint,
// we will fail to set the remote description.
TEST_P(WebRtcSessionTest, TestReceiveNonDtlsOfferWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
cricket::MediaSessionOptions options;
options.recv_video = true;
@ -1933,7 +1924,6 @@ TEST_P(WebRtcSessionTest, TestReceiveNonDtlsOfferWhenDtlsOn) {
// Test that we return a failure when applying a local answer that doesn't have
// a DTLS fingerprint when DTLS is required.
TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
@ -1949,7 +1939,6 @@ TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) {
// Test that we return a failure when applying a remote answer that doesn't have
// a DTLS fingerprint when DTLS is required.
TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
@ -3926,8 +3915,6 @@ TEST_F(WebRtcSessionTest, TestRtpDataChannel) {
}
TEST_P(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
configuration_.enable_rtp_data_channel = true;
options_.disable_sctp_data_channels = false;
@ -3940,7 +3927,6 @@ TEST_P(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
// Test that sctp_content_name/sctp_transport_name (used for stats) are correct
// before and after BUNDLE is negotiated.
TEST_P(WebRtcSessionTest, SctpContentAndTransportName) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetFactoryDtlsSrtp();
InitWithDtls(GetParam());
@ -3974,8 +3960,6 @@ TEST_P(WebRtcSessionTest, SctpContentAndTransportName) {
}
TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
@ -3984,7 +3968,6 @@ TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
}
TEST_P(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetFactoryDtlsSrtp();
InitWithDtls(GetParam());
@ -4016,8 +3999,6 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) {
// Test that if DTLS is enabled, we end up with an SctpTransport created
// (and not an RtpDataChannel).
TEST_P(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetLocalDescriptionWithDataChannel();
@ -4028,7 +4009,6 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
// Test that if SCTP is disabled, we don't end up with an SctpTransport
// created (or an RtpDataChannel).
TEST_P(WebRtcSessionTest, TestDisableSctpDataChannels) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
options_.disable_sctp_data_channels = true;
InitWithDtls(GetParam());
@ -4038,7 +4018,6 @@ TEST_P(WebRtcSessionTest, TestDisableSctpDataChannels) {
}
TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
const int new_send_port = 9998;
const int new_recv_port = 7775;
@ -4080,8 +4059,6 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
// WebRtcSession signals the SctpTransport creation request with the expected
// config.
TEST_P(WebRtcSessionTest, TestSctpDataChannelOpenMessage) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetLocalDescriptionWithDataChannel();
@ -4121,7 +4098,6 @@ TEST_P(WebRtcSessionTest, TestUsesProvidedCertificate) {
// identity generation is finished (even if a certificate is provided this is
// an async op).
TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
EXPECT_TRUE(session_->waiting_for_certificate_for_testing());
@ -4137,7 +4113,6 @@ TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
// identity generation is finished (even if a certificate is provided this is
// an async op).
TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@ -4158,7 +4133,6 @@ TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
// identity generation is finished (even if a certificate is provided this is
// an async op).
TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
@ -4170,7 +4144,6 @@ TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
// Verifies that CreateOffer fails when CreateOffer is called after async
// identity generation fails.
TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtlsIdentityGenFail();
EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
@ -4183,7 +4156,6 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
// before async identity generation is finished.
TEST_P(WebRtcSessionTest,
TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(GetParam(),
CreateSessionDescriptionRequest::kOffer);
}
@ -4192,7 +4164,6 @@ TEST_P(WebRtcSessionTest,
// before async identity generation fails.
TEST_F(WebRtcSessionTest,
TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
CreateSessionDescriptionRequest::kOffer);
}
@ -4201,7 +4172,6 @@ TEST_F(WebRtcSessionTest,
// before async identity generation is finished.
TEST_P(WebRtcSessionTest,
TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(
GetParam(), CreateSessionDescriptionRequest::kAnswer);
}
@ -4210,7 +4180,6 @@ TEST_P(WebRtcSessionTest,
// before async identity generation fails.
TEST_F(WebRtcSessionTest,
TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
CreateSessionDescriptionRequest::kAnswer);
}
@ -4254,7 +4223,6 @@ TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
// Tests that we can renegotiate new media content with ICE candidates in the
// new remote SDP.
TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@ -4284,7 +4252,6 @@ TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
// Tests that we can renegotiate new media content with ICE candidates separated
// from the remote SDP.
TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();