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/*
* 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 <cstdint>
#include <cstring>
#include <limits>
#include <vector>
#include "media/base/fake_rtp.h"
#include "pc/test/srtp_test_util.h"
#include "rtc_base/buffer.h"
#include "rtc_base/byte_order.h"
#include "rtc_base/copy_on_write_buffer.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);
rtp_packet_.EnsureCapacity(rtp_len_ + 10);
rtp_packet_.SetData(kPcmuFrame, rtp_len_);
rtcp_packet_.EnsureCapacity(rtcp_len_ + 4 + 10);
rtcp_packet_.SetData(kRtcpReport, rtcp_len_);
}
void TestProtectRtp(int crypto_suite) {
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
EXPECT_EQ(rtp_packet_.size(), rtp_len_ + rtp_auth_tag_len(crypto_suite));
// Check that Protect changed the content (up to the original length).
EXPECT_NE(0, std::memcmp(kPcmuFrame, rtp_packet_.data(), rtp_len_));
rtp_len_ = rtp_packet_.size();
}
void TestProtectRtcp(int crypto_suite) {
EXPECT_TRUE(s1_.ProtectRtcp(rtcp_packet_));
EXPECT_EQ(rtcp_packet_.size(),
rtcp_len_ + 4 + rtcp_auth_tag_len(crypto_suite));
// Check that Protect changed the content (up to the original length).
EXPECT_NE(0, std::memcmp(kRtcpReport, rtcp_packet_.data(), rtcp_len_));
rtcp_len_ = rtcp_packet_.size();
}
void TestUnprotectRtp(int crypto_suite) {
EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_));
EXPECT_EQ(rtp_packet_.size(), sizeof(kPcmuFrame));
EXPECT_EQ(0,
std::memcmp(kPcmuFrame, rtp_packet_.data(), rtp_packet_.size()));
}
void TestUnprotectRtcp(int crypto_suite) {
EXPECT_TRUE(s2_.UnprotectRtcp(rtcp_packet_));
EXPECT_EQ(rtcp_packet_.size(), sizeof(kRtcpReport));
EXPECT_EQ(
0, std::memcmp(kRtcpReport, rtcp_packet_.data(), rtcp_packet_.size()));
}
webrtc::test::ScopedKeyValueConfig field_trials_;
cricket::SrtpSession s1_;
cricket::SrtpSession s2_;
rtc::CopyOnWriteBuffer rtp_packet_;
rtc::CopyOnWriteBuffer rtcp_packet_;
size_t rtp_len_;
size_t rtcp_len_;
};
// Test that we can set up the session and keys properly.
TEST_F(SrtpSessionTest, TestGoodSetup) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
}
// Test that we can't change the keys once set.
TEST_F(SrtpSessionTest, TestBadSetup) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey2,
kEncryptedHeaderExtensionIds));
EXPECT_FALSE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey2,
kEncryptedHeaderExtensionIds));
}
// Test that we fail keys of the wrong length.
TEST_F(SrtpSessionTest, TestKeysTooShort) {
EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80,
rtc::ZeroOnFreeBuffer<uint8_t>(kTestKey1.data(), 1),
kEncryptedHeaderExtensionIds));
EXPECT_FALSE(s2_.SetReceive(
kSrtpAes128CmSha1_80, rtc::ZeroOnFreeBuffer<uint8_t>(kTestKey1.data(), 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,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
TestProtectRtp(kSrtpAes128CmSha1_80);
TestProtectRtcp(kSrtpAes128CmSha1_80);
TestUnprotectRtp(kSrtpAes128CmSha1_80);
TestUnprotectRtcp(kSrtpAes128CmSha1_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,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_32, kTestKey1,
kEncryptedHeaderExtensionIds));
TestProtectRtp(kSrtpAes128CmSha1_32);
TestProtectRtcp(kSrtpAes128CmSha1_32);
TestUnprotectRtp(kSrtpAes128CmSha1_32);
TestUnprotectRtcp(kSrtpAes128CmSha1_32);
}
TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1,
kEncryptedHeaderExtensionIds));
int64_t index;
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_, &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) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
TestProtectRtp(kSrtpAes128CmSha1_80);
rtp_packet_.MutableData<uint8_t>()[0] = 0x12;
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_));
EXPECT_METRIC_THAT(
webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"),
ElementsAre(Pair(srtp_err_status_bad_param, 1)));
TestProtectRtcp(kSrtpAes128CmSha1_80);
rtcp_packet_.MutableData<uint8_t>()[1] = 0x34;
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_));
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) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_));
EXPECT_METRIC_THAT(
webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"),
ElementsAre(Pair(srtp_err_status_auth_fail, 1)));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_));
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) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
// This buffer does not have extra capacity which we treat as an error.
rtc::CopyOnWriteBuffer rtp_packet(rtp_packet_.data(), rtp_packet_.size(),
rtp_packet_.size());
EXPECT_FALSE(s1_.ProtectRtp(rtp_packet));
// This buffer does not have extra capacity which we treat as an error.
rtc::CopyOnWriteBuffer rtcp_packet(rtcp_packet_.data(), rtcp_packet_.size(),
rtcp_packet_.size());
EXPECT_FALSE(s1_.ProtectRtcp(rtcp_packet));
}
TEST_F(SrtpSessionTest, TestReplay) {
static const uint16_t kMaxSeqnum = std::numeric_limits<uint16_t>::max() - 1;
static const uint16_t seqnum_big = 62275;
static const uint16_t seqnum_small = 10;
static const uint16_t replay_window = 1024;
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
// Initial sequence number.
SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum_big);
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
// Replay within the 1024 window should succeed.
SetBE16(rtp_packet_.MutableData<uint8_t>() + 2,
seqnum_big - replay_window + 1);
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
// Replay out side of the 1024 window should fail.
SetBE16(rtp_packet_.MutableData<uint8_t>() + 2,
seqnum_big - replay_window - 1);
EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_));
rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
// Increment sequence number to a small number.
SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum_small);
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
// Replay around 0 but out side of the 1024 window should fail.
SetBE16(rtp_packet_.MutableData<uint8_t>() + 2,
kMaxSeqnum + seqnum_small - replay_window - 1);
EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_));
rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
// Replay around 0 but within the 1024 window should succeed.
for (uint16_t seqnum = 65000; seqnum < 65003; ++seqnum) {
SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum);
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
}
// 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(rtp_packet_.MutableData<uint8_t>() + 2, seqnum_small + 1);
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
}
TEST_F(SrtpSessionTest, RemoveSsrc) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
// Encrypt and decrypt the packet once.
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_));
EXPECT_EQ(sizeof(kPcmuFrame), rtp_packet_.size());
EXPECT_EQ(0, std::memcmp(kPcmuFrame, rtp_packet_.data(), rtp_packet_.size()));
// Recreate the original packet and encrypt again.
rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
// Attempting to decrypt will fail as a replay attack.
// (srtp_err_status_replay_fail) since the sequence number was already seen.
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_));
// Remove the fake packet SSRC 1 from the session.
EXPECT_TRUE(s2_.RemoveSsrcFromSession(1));
EXPECT_FALSE(s2_.RemoveSsrcFromSession(1));
// Since the SRTP state was discarded, this is no longer a replay attack.
EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_));
EXPECT_EQ(sizeof(kPcmuFrame), rtp_packet_.size());
EXPECT_EQ(0, std::memcmp(kPcmuFrame, rtp_packet_.data(), rtp_packet_.size()));
EXPECT_TRUE(s2_.RemoveSsrcFromSession(1));
}
TEST_F(SrtpSessionTest, ProtectUnprotectWrapAroundRocMismatch) {
// This unit tests demonstrates why you should be careful when
// choosing the initial RTP sequence number as there can be decryption
// failures when it wraps around with packet loss. Pick your starting
// sequence number in the lower half of the range for robustness reasons,
// see packet_sequencer.cc for the code doing so.
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
// Buffers include enough room for the 10 byte SRTP auth tag so we can
// encrypt in place.
unsigned char kFrame1[] = {
// clang-format off
// PT=0, SN=65535, TS=0, SSRC=1
0x80, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
0xBE, 0xEF, // data bytes
// Space for the SRTP auth tag
0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
// clang-format on
};
rtc::CopyOnWriteBuffer packet1(kFrame1, sizeof(kFrame1) - 10,
sizeof(kFrame1));
unsigned char kFrame2[] = {
// clang-format off
// PT=0, SN=1, TS=0, SSRC=1
0x80, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
0xBE, 0xEF, // data bytes
// Space for the SRTP auth tag
0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
// clang-format on
};
rtc::CopyOnWriteBuffer packet2(kFrame2, sizeof(kFrame2) - 10,
sizeof(kFrame1));
const unsigned char kPayload[] = {0xBE, 0xEF};
// Encrypt the frames in-order. There is a sequence number rollover from
// 65535 to 1 (skipping 0) and the second packet gets encrypted with a
// roll-over counter (ROC) of 1. See
// https://datatracker.ietf.org/doc/html/rfc3711#section-3.3.1
EXPECT_TRUE(s1_.ProtectRtp(packet1));
EXPECT_EQ(packet1.size(), 24u);
EXPECT_TRUE(s1_.ProtectRtp(packet2));
EXPECT_EQ(packet2.size(), 24u);
// If we decrypt frame 2 first it will have a ROC of 1 but the receiver
// does not know this is a rollover so will attempt with a ROC of 0.
// Note: If libsrtp is modified to attempt to decrypt with ROC=1 for this
// case, this test will fail and needs to be modified accordingly to unblock
// the roll. See https://issues.webrtc.org/353565743 for details.
EXPECT_FALSE(s2_.UnprotectRtp(packet2));
// Decrypt frame 1.
EXPECT_TRUE(s2_.UnprotectRtp(packet1));
ASSERT_EQ(packet1.size(), 14u);
EXPECT_EQ(0, std::memcmp(packet1.data() + 12, kPayload, sizeof(kPayload)));
// Now decrypt frame 2 again. A rollover is detected which increases
// the ROC to 1 so this succeeds.
EXPECT_TRUE(s2_.UnprotectRtp(packet2));
ASSERT_EQ(packet2.size(), 14u);
EXPECT_EQ(0, std::memcmp(packet2.data() + 12, kPayload, sizeof(kPayload)));
}
TEST_F(SrtpSessionTest, ProtectGetPacketIndex) {
EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1,
kEncryptedHeaderExtensionIds));
// Buffers include enough room for the 10 byte SRTP auth tag so we can
// encrypt in place.
unsigned char kFrame1[] = {
// clang-format off
// PT=0, SN=65535, TS=0, SSRC=1
0x80, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
0xBE, 0xEF, // data bytes
// Space for the SRTP auth tag
0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
// clang-format on
};
rtc::CopyOnWriteBuffer packet1(kFrame1, sizeof(kFrame1) - 10,
sizeof(kFrame1));
unsigned char kFrame2[] = {
// clang-format off
// PT=0, SN=1, TS=0, SSRC=1
0x80, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
0xBE, 0xEF, // data bytes
// Space for the SRTP auth tag
0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
// clang-format on
};
rtc::CopyOnWriteBuffer packet2(kFrame2, sizeof(kFrame2) - 10,
sizeof(kFrame1));
// Encrypt the frames in-order. There is a sequence number rollover from
// 65535 to 1 (skipping 0) and the second packet gets encrypted with a
// roll-over counter (ROC) of 1. See
// https://datatracker.ietf.org/doc/html/rfc3711#section-3.3.1
int64_t index;
EXPECT_TRUE(s1_.ProtectRtp(packet1, &index));
EXPECT_EQ(packet1.size(), 24u);
EXPECT_EQ(index, 0xffff00000000); // ntohl(65535 << 16)
EXPECT_TRUE(s1_.ProtectRtp(packet2, &index));
EXPECT_EQ(packet2.size(), 24u);
EXPECT_EQ(index, 0x10001000000); // ntohl(65537 << 16)
}
} // namespace rtc