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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/srtp_session_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/srtp_session_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/srtp_session_unittest.cc | 254 |
1 files changed, 254 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/srtp_session_unittest.cc b/third_party/libwebrtc/pc/srtp_session_unittest.cc new file mode 100644 index 0000000000..16a840a307 --- /dev/null +++ b/third_party/libwebrtc/pc/srtp_session_unittest.cc @@ -0,0 +1,254 @@ +/* + * Copyright 2004 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "pc/srtp_session.h" + +#include <string.h> + +#include <string> + +#include "media/base/fake_rtp.h" +#include "pc/test/srtp_test_util.h" +#include "rtc_base/byte_order.h" +#include "rtc_base/ssl_stream_adapter.h" // For rtc::SRTP_* +#include "system_wrappers/include/metrics.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/scoped_key_value_config.h" +#include "third_party/libsrtp/include/srtp.h" + +using ::testing::ElementsAre; +using ::testing::Pair; + +namespace rtc { + +std::vector<int> kEncryptedHeaderExtensionIds; + +class SrtpSessionTest : public ::testing::Test { + public: + SrtpSessionTest() : s1_(field_trials_), s2_(field_trials_) { + webrtc::metrics::Reset(); + } + + protected: + virtual void SetUp() { + rtp_len_ = sizeof(kPcmuFrame); + rtcp_len_ = sizeof(kRtcpReport); + memcpy(rtp_packet_, kPcmuFrame, rtp_len_); + memcpy(rtcp_packet_, kRtcpReport, rtcp_len_); + } + void TestProtectRtp(const std::string& cs) { + int out_len = 0; + EXPECT_TRUE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + EXPECT_EQ(out_len, rtp_len_ + rtp_auth_tag_len(cs)); + EXPECT_NE(0, memcmp(rtp_packet_, kPcmuFrame, rtp_len_)); + rtp_len_ = out_len; + } + void TestProtectRtcp(const std::string& cs) { + int out_len = 0; + EXPECT_TRUE(s1_.ProtectRtcp(rtcp_packet_, rtcp_len_, sizeof(rtcp_packet_), + &out_len)); + EXPECT_EQ(out_len, rtcp_len_ + 4 + rtcp_auth_tag_len(cs)); // NOLINT + EXPECT_NE(0, memcmp(rtcp_packet_, kRtcpReport, rtcp_len_)); + rtcp_len_ = out_len; + } + void TestUnprotectRtp(const std::string& cs) { + int out_len = 0, expected_len = sizeof(kPcmuFrame); + EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len)); + EXPECT_EQ(expected_len, out_len); + EXPECT_EQ(0, memcmp(rtp_packet_, kPcmuFrame, out_len)); + } + void TestUnprotectRtcp(const std::string& cs) { + int out_len = 0, expected_len = sizeof(kRtcpReport); + EXPECT_TRUE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len)); + EXPECT_EQ(expected_len, out_len); + EXPECT_EQ(0, memcmp(rtcp_packet_, kRtcpReport, out_len)); + } + webrtc::test::ScopedKeyValueConfig field_trials_; + cricket::SrtpSession s1_; + cricket::SrtpSession s2_; + char rtp_packet_[sizeof(kPcmuFrame) + 10]; + char rtcp_packet_[sizeof(kRtcpReport) + 4 + 10]; + int rtp_len_; + int rtcp_len_; +}; + +// Test that we can set up the session and keys properly. +TEST_F(SrtpSessionTest, TestGoodSetup) { + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); +} + +// Test that we can't change the keys once set. +TEST_F(SrtpSessionTest, TestBadSetup) { + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey2, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey2, kTestKeyLen, + kEncryptedHeaderExtensionIds)); +} + +// Test that we fail keys of the wrong length. +TEST_F(SrtpSessionTest, TestKeysTooShort) { + EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, 1, + kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, 1, + kEncryptedHeaderExtensionIds)); +} + +// Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_80. +TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_80) { + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + TestProtectRtp(kCsAesCm128HmacSha1_80); + TestProtectRtcp(kCsAesCm128HmacSha1_80); + TestUnprotectRtp(kCsAesCm128HmacSha1_80); + TestUnprotectRtcp(kCsAesCm128HmacSha1_80); +} + +// Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_32. +TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_32) { + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_32, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + TestProtectRtp(kCsAesCm128HmacSha1_32); + TestProtectRtcp(kCsAesCm128HmacSha1_32); + TestUnprotectRtp(kCsAesCm128HmacSha1_32); + TestUnprotectRtcp(kCsAesCm128HmacSha1_32); +} + +TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) { + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + int64_t index; + int out_len = 0; + EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), + &out_len, &index)); + // `index` will be shifted by 16. + int64_t be64_index = static_cast<int64_t>(NetworkToHost64(1 << 16)); + EXPECT_EQ(be64_index, index); +} + +// Test that we fail to unprotect if someone tampers with the RTP/RTCP paylaods. +TEST_F(SrtpSessionTest, TestTamperReject) { + int out_len; + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + TestProtectRtp(kCsAesCm128HmacSha1_80); + TestProtectRtcp(kCsAesCm128HmacSha1_80); + rtp_packet_[0] = 0x12; + rtcp_packet_[1] = 0x34; + EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len)); + EXPECT_METRIC_THAT( + webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"), + ElementsAre(Pair(srtp_err_status_bad_param, 1))); + EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len)); + EXPECT_METRIC_THAT( + webrtc::metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"), + ElementsAre(Pair(srtp_err_status_auth_fail, 1))); +} + +// Test that we fail to unprotect if the payloads are not authenticated. +TEST_F(SrtpSessionTest, TestUnencryptReject) { + int out_len; + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len)); + EXPECT_METRIC_THAT( + webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"), + ElementsAre(Pair(srtp_err_status_auth_fail, 1))); + EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len)); + EXPECT_METRIC_THAT( + webrtc::metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"), + ElementsAre(Pair(srtp_err_status_cant_check, 1))); +} + +// Test that we fail when using buffers that are too small. +TEST_F(SrtpSessionTest, TestBuffersTooSmall) { + int out_len; + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_) - 10, + &out_len)); + EXPECT_FALSE(s1_.ProtectRtcp(rtcp_packet_, rtcp_len_, + sizeof(rtcp_packet_) - 14, &out_len)); +} + +TEST_F(SrtpSessionTest, TestReplay) { + static const uint16_t kMaxSeqnum = static_cast<uint16_t>(-1); + static const uint16_t seqnum_big = 62275; + static const uint16_t seqnum_small = 10; + static const uint16_t replay_window = 1024; + int out_len; + + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen, + kEncryptedHeaderExtensionIds)); + + // Initial sequence number. + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_big); + EXPECT_TRUE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + + // Replay within the 1024 window should succeed. + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, + seqnum_big - replay_window + 1); + EXPECT_TRUE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + + // Replay out side of the 1024 window should fail. + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, + seqnum_big - replay_window - 1); + EXPECT_FALSE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + + // Increment sequence number to a small number. + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_small); + EXPECT_TRUE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + + // Replay around 0 but out side of the 1024 window should fail. + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, + kMaxSeqnum + seqnum_small - replay_window - 1); + EXPECT_FALSE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + + // Replay around 0 but within the 1024 window should succeed. + for (uint16_t seqnum = 65000; seqnum < 65003; ++seqnum) { + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum); + EXPECT_TRUE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); + } + + // Go back to normal sequence nubmer. + // NOTE: without the fix in libsrtp, this would fail. This is because + // without the fix, the loop above would keep incrementing local sequence + // number in libsrtp, eventually the new sequence number would go out side + // of the window. + SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_small + 1); + EXPECT_TRUE( + s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len)); +} + +} // namespace rtc |