(3) Rename files to snake_case: move the files
Mechanically generated with this command: tools_webrtc/do-rename.sh move all-renames.txt Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690 Reviewed-on: https://webrtc-review.googlesource.com/c/115481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26225}
This commit is contained in:
Steve Anton
253
pc/srtp_session_unittest.cc
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253
pc/srtp_session_unittest.cc
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/*
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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/srtpsession.h"
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#include <string.h>
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#include <string>
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#include "media/base/fakertp.h"
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#include "pc/srtptestutil.h"
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#include "rtc_base/byteorder.h"
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#include "rtc_base/sslstreamadapter.h" // For rtc::SRTP_*
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#include "system_wrappers/include/metrics.h"
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#include "test/gtest.h"
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#include "third_party/libsrtp/include/srtp.h"
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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() { webrtc::metrics::Reset(); }
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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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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(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_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(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s1_.SetSend(SRTP_AES128_CM_SHA1_80, kTestKey2, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s2_.SetRecv(SRTP_AES128_CM_SHA1_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(SRTP_AES128_CM_SHA1_80, kTestKey1, 1,
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kEncryptedHeaderExtensionIds));
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EXPECT_FALSE(s2_.SetRecv(SRTP_AES128_CM_SHA1_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(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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TestProtectRtp(CS_AES_CM_128_HMAC_SHA1_80);
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TestProtectRtcp(CS_AES_CM_128_HMAC_SHA1_80);
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TestUnprotectRtp(CS_AES_CM_128_HMAC_SHA1_80);
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TestUnprotectRtcp(CS_AES_CM_128_HMAC_SHA1_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(SRTP_AES128_CM_SHA1_32, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_32, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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TestProtectRtp(CS_AES_CM_128_HMAC_SHA1_32);
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TestProtectRtcp(CS_AES_CM_128_HMAC_SHA1_32);
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TestUnprotectRtp(CS_AES_CM_128_HMAC_SHA1_32);
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TestUnprotectRtcp(CS_AES_CM_128_HMAC_SHA1_32);
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}
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TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) {
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EXPECT_TRUE(s1_.SetSend(SRTP_AES128_CM_SHA1_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(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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TestProtectRtp(CS_AES_CM_128_HMAC_SHA1_80);
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TestProtectRtcp(CS_AES_CM_128_HMAC_SHA1_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_EQ(1, webrtc::metrics::NumSamples(
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"WebRTC.PeerConnection.SrtpUnprotectError"));
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EXPECT_EQ(
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1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
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srtp_err_status_bad_param));
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EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
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EXPECT_EQ(1, webrtc::metrics::NumSamples(
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"WebRTC.PeerConnection.SrtcpUnprotectError"));
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EXPECT_EQ(
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1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
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srtp_err_status_auth_fail));
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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(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_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_EQ(1, webrtc::metrics::NumSamples(
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"WebRTC.PeerConnection.SrtpUnprotectError"));
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EXPECT_EQ(
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1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
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srtp_err_status_auth_fail));
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EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
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EXPECT_EQ(1, webrtc::metrics::NumSamples(
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"WebRTC.PeerConnection.SrtcpUnprotectError"));
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EXPECT_EQ(
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1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
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srtp_err_status_cant_check));
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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(SRTP_AES128_CM_SHA1_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(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
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kEncryptedHeaderExtensionIds));
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EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_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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} // namespace rtc
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