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https://github.com/mollyim/webrtc.git
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977b56c9e9
This uses libSRTPs srtp_remove_stream() https://github.com/cisco/libsrtp/blob/main/include/srtp.h#L597 method to remove SSRCs from the libSRTP session when they are removed from the RTP demuxer. This works even when the stream was added automatically via the ssrc_any_inbound mechanism. Only streams for inbound SSRCs that were added explicitly via SDP negotiation are removed. Guarded by WebRTC-SrtpRemoveReceiveStream field trial. BUG=webrtc:15604 Change-Id: I655bde5f8ddf26ac91395ef54bd1b3c598813380 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324720 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#41105}
286 lines
12 KiB
C++
286 lines
12 KiB
C++
/*
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* Copyright 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/srtp_session.h"
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#include <string.h>
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#include <string>
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#include "media/base/fake_rtp.h"
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#include "pc/test/srtp_test_util.h"
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#include "rtc_base/byte_order.h"
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#include "rtc_base/ssl_stream_adapter.h" // For rtc::SRTP_*
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#include "system_wrappers/include/metrics.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/scoped_key_value_config.h"
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#include "third_party/libsrtp/include/srtp.h"
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using ::testing::ElementsAre;
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using ::testing::Pair;
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namespace rtc {
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std::vector<int> kEncryptedHeaderExtensionIds;
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class SrtpSessionTest : public ::testing::Test {
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public:
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SrtpSessionTest() : s1_(field_trials_), s2_(field_trials_) {
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webrtc::metrics::Reset();
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}
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protected:
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virtual void SetUp() {
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rtp_len_ = sizeof(kPcmuFrame);
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rtcp_len_ = sizeof(kRtcpReport);
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memcpy(rtp_packet_, kPcmuFrame, rtp_len_);
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memcpy(rtcp_packet_, kRtcpReport, rtcp_len_);
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}
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void TestProtectRtp(const std::string& cs) {
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int out_len = 0;
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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EXPECT_EQ(out_len, rtp_len_ + rtp_auth_tag_len(cs));
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EXPECT_NE(0, memcmp(rtp_packet_, kPcmuFrame, rtp_len_));
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rtp_len_ = out_len;
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}
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void TestProtectRtcp(const std::string& cs) {
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int out_len = 0;
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EXPECT_TRUE(s1_.ProtectRtcp(rtcp_packet_, rtcp_len_, sizeof(rtcp_packet_),
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&out_len));
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EXPECT_EQ(out_len, rtcp_len_ + 4 + rtcp_auth_tag_len(cs)); // NOLINT
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EXPECT_NE(0, memcmp(rtcp_packet_, kRtcpReport, rtcp_len_));
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rtcp_len_ = out_len;
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}
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void TestUnprotectRtp(const std::string& cs) {
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int out_len = 0, expected_len = sizeof(kPcmuFrame);
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EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
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EXPECT_EQ(expected_len, out_len);
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EXPECT_EQ(0, memcmp(rtp_packet_, kPcmuFrame, out_len));
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}
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void TestUnprotectRtcp(const std::string& cs) {
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int out_len = 0, expected_len = sizeof(kRtcpReport);
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EXPECT_TRUE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
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EXPECT_EQ(expected_len, out_len);
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EXPECT_EQ(0, memcmp(rtcp_packet_, kRtcpReport, out_len));
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}
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webrtc::test::ScopedKeyValueConfig field_trials_;
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cricket::SrtpSession s1_;
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cricket::SrtpSession s2_;
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char rtp_packet_[sizeof(kPcmuFrame) + 10];
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char rtcp_packet_[sizeof(kRtcpReport) + 4 + 10];
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int rtp_len_;
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int rtcp_len_;
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};
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// Test that we can set up the session and keys properly.
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TEST_F(SrtpSessionTest, TestGoodSetup) {
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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}
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// Test that we can't change the keys once set.
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TEST_F(SrtpSessionTest, TestBadSetup) {
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey2, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey2, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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}
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// Test that we fail keys of the wrong length.
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TEST_F(SrtpSessionTest, TestKeysTooShort) {
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EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, 1,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, 1,
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kEncryptedHeaderExtensionIds));
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}
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// Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_80.
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TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_80) {
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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TestProtectRtp(kCsAesCm128HmacSha1_80);
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TestProtectRtcp(kCsAesCm128HmacSha1_80);
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TestUnprotectRtp(kCsAesCm128HmacSha1_80);
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TestUnprotectRtcp(kCsAesCm128HmacSha1_80);
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}
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// Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_32.
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TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_32) {
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_32, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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TestProtectRtp(kCsAesCm128HmacSha1_32);
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TestProtectRtcp(kCsAesCm128HmacSha1_32);
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TestUnprotectRtp(kCsAesCm128HmacSha1_32);
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TestUnprotectRtcp(kCsAesCm128HmacSha1_32);
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}
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TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) {
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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int64_t index;
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int out_len = 0;
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EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_),
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&out_len, &index));
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// `index` will be shifted by 16.
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int64_t be64_index = static_cast<int64_t>(NetworkToHost64(1 << 16));
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EXPECT_EQ(be64_index, index);
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}
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// Test that we fail to unprotect if someone tampers with the RTP/RTCP paylaods.
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TEST_F(SrtpSessionTest, TestTamperReject) {
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int out_len;
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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TestProtectRtp(kCsAesCm128HmacSha1_80);
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TestProtectRtcp(kCsAesCm128HmacSha1_80);
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rtp_packet_[0] = 0x12;
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rtcp_packet_[1] = 0x34;
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EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
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EXPECT_METRIC_THAT(
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webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"),
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ElementsAre(Pair(srtp_err_status_bad_param, 1)));
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EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
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EXPECT_METRIC_THAT(
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webrtc::metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"),
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ElementsAre(Pair(srtp_err_status_auth_fail, 1)));
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}
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// Test that we fail to unprotect if the payloads are not authenticated.
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TEST_F(SrtpSessionTest, TestUnencryptReject) {
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int out_len;
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
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EXPECT_METRIC_THAT(
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webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"),
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ElementsAre(Pair(srtp_err_status_auth_fail, 1)));
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EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
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EXPECT_METRIC_THAT(
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webrtc::metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"),
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ElementsAre(Pair(srtp_err_status_cant_check, 1)));
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}
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// Test that we fail when using buffers that are too small.
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TEST_F(SrtpSessionTest, TestBuffersTooSmall) {
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int out_len;
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_) - 10,
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&out_len));
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EXPECT_FALSE(s1_.ProtectRtcp(rtcp_packet_, rtcp_len_,
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sizeof(rtcp_packet_) - 14, &out_len));
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}
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TEST_F(SrtpSessionTest, TestReplay) {
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static const uint16_t kMaxSeqnum = static_cast<uint16_t>(-1);
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static const uint16_t seqnum_big = 62275;
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static const uint16_t seqnum_small = 10;
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static const uint16_t replay_window = 1024;
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int out_len;
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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// Initial sequence number.
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_big);
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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// Replay within the 1024 window should succeed.
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2,
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seqnum_big - replay_window + 1);
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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// Replay out side of the 1024 window should fail.
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2,
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seqnum_big - replay_window - 1);
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EXPECT_FALSE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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// Increment sequence number to a small number.
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_small);
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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// Replay around 0 but out side of the 1024 window should fail.
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2,
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kMaxSeqnum + seqnum_small - replay_window - 1);
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EXPECT_FALSE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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// Replay around 0 but within the 1024 window should succeed.
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for (uint16_t seqnum = 65000; seqnum < 65003; ++seqnum) {
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum);
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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}
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// Go back to normal sequence nubmer.
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// NOTE: without the fix in libsrtp, this would fail. This is because
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// without the fix, the loop above would keep incrementing local sequence
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// number in libsrtp, eventually the new sequence number would go out side
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// of the window.
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SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_small + 1);
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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}
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TEST_F(SrtpSessionTest, RemoveSsrc) {
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EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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int out_len;
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// Encrypt and decrypt the packet once.
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_, out_len, &out_len));
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EXPECT_EQ(rtp_len_, out_len);
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EXPECT_EQ(0, memcmp(rtp_packet_, kPcmuFrame, out_len));
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// Recreate the original packet and encrypt again.
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memcpy(rtp_packet_, kPcmuFrame, rtp_len_);
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EXPECT_TRUE(
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s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
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// Attempting to decrypt will fail as a replay attack.
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// (srtp_err_status_replay_fail) since the sequence number was already seen.
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EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, out_len, &out_len));
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// Remove the fake packet SSRC 1 from the session.
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EXPECT_TRUE(s2_.RemoveSsrcFromSession(1));
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EXPECT_FALSE(s2_.RemoveSsrcFromSession(1));
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// Since the SRTP state was discarded, this is no longer a replay attack.
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EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_, out_len, &out_len));
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EXPECT_EQ(rtp_len_, out_len);
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EXPECT_EQ(0, memcmp(rtp_packet_, kPcmuFrame, out_len));
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EXPECT_TRUE(s2_.RemoveSsrcFromSession(1));
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}
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} // namespace rtc
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