blob: 76ab3a8fe80690f57b3afc416edae219f09504aa [file] [log] [blame]
/*
* Copyright 2017 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 <iomanip>
#include "absl/base/attributes.h"
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "pc/external_hmac.h"
#include "rtc_base/logging.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "third_party/libsrtp/include/srtp.h"
#include "third_party/libsrtp/include/srtp_priv.h"
namespace cricket {
using ::webrtc::ParseRtpSequenceNumber;
// One more than the maximum libsrtp error code. Required by
// RTC_HISTOGRAM_ENUMERATION. Keep this in sync with srtp_error_status_t defined
// in srtp.h.
constexpr int kSrtpErrorCodeBoundary = 28;
SrtpSession::SrtpSession() {
dump_plain_rtp_ = webrtc::field_trial::IsEnabled("WebRTC-Debugging-RtpDump");
}
SrtpSession::~SrtpSession() {
if (session_) {
srtp_set_user_data(session_, nullptr);
srtp_dealloc(session_);
}
if (inited_) {
DecrementLibsrtpUsageCountAndMaybeDeinit();
}
}
bool SrtpSession::SetSend(int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
return SetKey(ssrc_any_outbound, cs, key, len, extension_ids);
}
bool SrtpSession::UpdateSend(int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
return UpdateKey(ssrc_any_outbound, cs, key, len, extension_ids);
}
bool SrtpSession::SetRecv(int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
return SetKey(ssrc_any_inbound, cs, key, len, extension_ids);
}
bool SrtpSession::UpdateRecv(int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
return UpdateKey(ssrc_any_inbound, cs, key, len, extension_ids);
}
bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
RTC_DCHECK(thread_checker_.IsCurrent());
if (!session_) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet: no SRTP Session";
return false;
}
// Note: the need_len differs from the libsrtp recommendatĐľon to ensure
// SRTP_MAX_TRAILER_LEN bytes of free space after the data. WebRTC
// never includes a MKI, therefore the amount of bytes added by the
// srtp_protect call is known in advance and depends on the cipher suite.
int need_len = in_len + rtp_auth_tag_len_; // NOLINT
if (max_len < need_len) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length "
<< max_len << " is less than the needed " << need_len;
return false;
}
if (dump_plain_rtp_) {
DumpPacket(p, in_len, /*outbound=*/true);
}
*out_len = in_len;
int err = srtp_protect(session_, p, out_len);
int seq_num = ParseRtpSequenceNumber(
rtc::MakeArrayView(reinterpret_cast<const uint8_t*>(p), in_len));
if (err != srtp_err_status_ok) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet, seqnum=" << seq_num
<< ", err=" << err
<< ", last seqnum=" << last_send_seq_num_;
return false;
}
last_send_seq_num_ = seq_num;
return true;
}
bool SrtpSession::ProtectRtp(void* p,
int in_len,
int max_len,
int* out_len,
int64_t* index) {
if (!ProtectRtp(p, in_len, max_len, out_len)) {
return false;
}
return (index) ? GetSendStreamPacketIndex(p, in_len, index) : true;
}
bool SrtpSession::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) {
RTC_DCHECK(thread_checker_.IsCurrent());
if (!session_) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet: no SRTP Session";
return false;
}
// Note: the need_len differs from the libsrtp recommendatĐľon to ensure
// SRTP_MAX_TRAILER_LEN bytes of free space after the data. WebRTC
// never includes a MKI, therefore the amount of bytes added by the
// srtp_protect_rtp call is known in advance and depends on the cipher suite.
int need_len = in_len + sizeof(uint32_t) + rtcp_auth_tag_len_; // NOLINT
if (max_len < need_len) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet: The buffer length "
<< max_len << " is less than the needed " << need_len;
return false;
}
if (dump_plain_rtp_) {
DumpPacket(p, in_len, /*outbound=*/true);
}
*out_len = in_len;
int err = srtp_protect_rtcp(session_, p, out_len);
if (err != srtp_err_status_ok) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet, err=" << err;
return false;
}
return true;
}
bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
RTC_DCHECK(thread_checker_.IsCurrent());
if (!session_) {
RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet: no SRTP Session";
return false;
}
*out_len = in_len;
int err = srtp_unprotect(session_, p, out_len);
if (err != srtp_err_status_ok) {
// Limit the error logging to avoid excessive logs when there are lots of
// bad packets.
const int kFailureLogThrottleCount = 100;
if (decryption_failure_count_ % kFailureLogThrottleCount == 0) {
RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err
<< ", previous failure count: "
<< decryption_failure_count_;
}
++decryption_failure_count_;
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError",
static_cast<int>(err), kSrtpErrorCodeBoundary);
return false;
}
if (dump_plain_rtp_) {
DumpPacket(p, *out_len, /*outbound=*/false);
}
return true;
}
bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
RTC_DCHECK(thread_checker_.IsCurrent());
if (!session_) {
RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet: no SRTP Session";
return false;
}
*out_len = in_len;
int err = srtp_unprotect_rtcp(session_, p, out_len);
if (err != srtp_err_status_ok) {
RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err;
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtcpUnprotectError",
static_cast<int>(err), kSrtpErrorCodeBoundary);
return false;
}
if (dump_plain_rtp_) {
DumpPacket(p, *out_len, /*outbound=*/false);
}
return true;
}
bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(IsExternalAuthActive());
if (!IsExternalAuthActive()) {
return false;
}
ExternalHmacContext* external_hmac = nullptr;
// stream_template will be the reference context for other streams.
// Let's use it for getting the keys.
srtp_stream_ctx_t* srtp_context = session_->stream_template;
if (srtp_context && srtp_context->session_keys &&
srtp_context->session_keys->rtp_auth) {
external_hmac = reinterpret_cast<ExternalHmacContext*>(
srtp_context->session_keys->rtp_auth->state);
}
if (!external_hmac) {
RTC_LOG(LS_ERROR) << "Failed to get auth keys from libsrtp!.";
return false;
}
*key = external_hmac->key;
*key_len = external_hmac->key_length;
*tag_len = rtp_auth_tag_len_;
return true;
}
int SrtpSession::GetSrtpOverhead() const {
return rtp_auth_tag_len_;
}
void SrtpSession::EnableExternalAuth() {
RTC_DCHECK(!session_);
external_auth_enabled_ = true;
}
bool SrtpSession::IsExternalAuthEnabled() const {
return external_auth_enabled_;
}
bool SrtpSession::IsExternalAuthActive() const {
return external_auth_active_;
}
bool SrtpSession::GetSendStreamPacketIndex(void* p,
int in_len,
int64_t* index) {
RTC_DCHECK(thread_checker_.IsCurrent());
srtp_hdr_t* hdr = reinterpret_cast<srtp_hdr_t*>(p);
srtp_stream_ctx_t* stream = srtp_get_stream(session_, hdr->ssrc);
if (!stream) {
return false;
}
// Shift packet index, put into network byte order
*index = static_cast<int64_t>(rtc::NetworkToHost64(
srtp_rdbx_get_packet_index(&stream->rtp_rdbx) << 16));
return true;
}
bool SrtpSession::DoSetKey(int type,
int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
RTC_DCHECK(thread_checker_.IsCurrent());
srtp_policy_t policy;
memset(&policy, 0, sizeof(policy));
if (!(srtp_crypto_policy_set_from_profile_for_rtp(
&policy.rtp, (srtp_profile_t)cs) == srtp_err_status_ok &&
srtp_crypto_policy_set_from_profile_for_rtcp(
&policy.rtcp, (srtp_profile_t)cs) == srtp_err_status_ok)) {
RTC_LOG(LS_ERROR) << "Failed to " << (session_ ? "update" : "create")
<< " SRTP session: unsupported cipher_suite " << cs;
return false;
}
if (!key || len != static_cast<size_t>(policy.rtp.cipher_key_len)) {
RTC_LOG(LS_ERROR) << "Failed to " << (session_ ? "update" : "create")
<< " SRTP session: invalid key";
return false;
}
policy.ssrc.type = static_cast<srtp_ssrc_type_t>(type);
policy.ssrc.value = 0;
policy.key = const_cast<uint8_t*>(key);
// TODO(astor) parse window size from WSH session-param
policy.window_size = 1024;
policy.allow_repeat_tx = 1;
// If external authentication option is enabled, supply custom auth module
// id EXTERNAL_HMAC_SHA1 in the policy structure.
// We want to set this option only for rtp packets.
// By default policy structure is initialized to HMAC_SHA1.
// Enable external HMAC authentication only for outgoing streams and only
// for cipher suites that support it (i.e. only non-GCM cipher suites).
if (type == ssrc_any_outbound && IsExternalAuthEnabled() &&
!rtc::IsGcmCryptoSuite(cs)) {
policy.rtp.auth_type = EXTERNAL_HMAC_SHA1;
}
if (!extension_ids.empty()) {
policy.enc_xtn_hdr = const_cast<int*>(&extension_ids[0]);
policy.enc_xtn_hdr_count = static_cast<int>(extension_ids.size());
}
policy.next = nullptr;
if (!session_) {
int err = srtp_create(&session_, &policy);
if (err != srtp_err_status_ok) {
session_ = nullptr;
RTC_LOG(LS_ERROR) << "Failed to create SRTP session, err=" << err;
return false;
}
srtp_set_user_data(session_, this);
} else {
int err = srtp_update(session_, &policy);
if (err != srtp_err_status_ok) {
RTC_LOG(LS_ERROR) << "Failed to update SRTP session, err=" << err;
return false;
}
}
rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
rtcp_auth_tag_len_ = policy.rtcp.auth_tag_len;
external_auth_active_ = (policy.rtp.auth_type == EXTERNAL_HMAC_SHA1);
return true;
}
bool SrtpSession::SetKey(int type,
int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
RTC_DCHECK(thread_checker_.IsCurrent());
if (session_) {
RTC_LOG(LS_ERROR) << "Failed to create SRTP session: "
"SRTP session already created";
return false;
}
// This is the first time we need to actually interact with libsrtp, so
// initialize it if needed.
if (IncrementLibsrtpUsageCountAndMaybeInit()) {
inited_ = true;
} else {
return false;
}
return DoSetKey(type, cs, key, len, extension_ids);
}
bool SrtpSession::UpdateKey(int type,
int cs,
const uint8_t* key,
size_t len,
const std::vector<int>& extension_ids) {
RTC_DCHECK(thread_checker_.IsCurrent());
if (!session_) {
RTC_LOG(LS_ERROR) << "Failed to update non-existing SRTP session";
return false;
}
return DoSetKey(type, cs, key, len, extension_ids);
}
ABSL_CONST_INIT int g_libsrtp_usage_count = 0;
ABSL_CONST_INIT webrtc::GlobalMutex g_libsrtp_lock(absl::kConstInit);
void ProhibitLibsrtpInitialization() {
webrtc::GlobalMutexLock ls(&g_libsrtp_lock);
++g_libsrtp_usage_count;
}
// static
bool SrtpSession::IncrementLibsrtpUsageCountAndMaybeInit() {
webrtc::GlobalMutexLock ls(&g_libsrtp_lock);
RTC_DCHECK_GE(g_libsrtp_usage_count, 0);
if (g_libsrtp_usage_count == 0) {
int err;
err = srtp_init();
if (err != srtp_err_status_ok) {
RTC_LOG(LS_ERROR) << "Failed to init SRTP, err=" << err;
return false;
}
err = srtp_install_event_handler(&SrtpSession::HandleEventThunk);
if (err != srtp_err_status_ok) {
RTC_LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err;
return false;
}
err = external_crypto_init();
if (err != srtp_err_status_ok) {
RTC_LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err;
return false;
}
}
++g_libsrtp_usage_count;
return true;
}
// static
void SrtpSession::DecrementLibsrtpUsageCountAndMaybeDeinit() {
webrtc::GlobalMutexLock ls(&g_libsrtp_lock);
RTC_DCHECK_GE(g_libsrtp_usage_count, 1);
if (--g_libsrtp_usage_count == 0) {
int err = srtp_shutdown();
if (err) {
RTC_LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err;
}
}
}
void SrtpSession::HandleEvent(const srtp_event_data_t* ev) {
RTC_DCHECK(thread_checker_.IsCurrent());
switch (ev->event) {
case event_ssrc_collision:
RTC_LOG(LS_INFO) << "SRTP event: SSRC collision";
break;
case event_key_soft_limit:
RTC_LOG(LS_INFO) << "SRTP event: reached soft key usage limit";
break;
case event_key_hard_limit:
RTC_LOG(LS_INFO) << "SRTP event: reached hard key usage limit";
break;
case event_packet_index_limit:
RTC_LOG(LS_INFO)
<< "SRTP event: reached hard packet limit (2^48 packets)";
break;
default:
RTC_LOG(LS_INFO) << "SRTP event: unknown " << ev->event;
break;
}
}
void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
// Callback will be executed from same thread that calls the "srtp_protect"
// and "srtp_unprotect" functions.
SrtpSession* session =
static_cast<SrtpSession*>(srtp_get_user_data(ev->session));
if (session) {
session->HandleEvent(ev);
}
}
// Logs the unencrypted packet in text2pcap format. This can then be
// extracted by searching for RTP_DUMP
// grep RTP_DUMP chrome_debug.log > in.txt
// and converted to pcap using
// text2pcap -D -u 1000,2000 -t %H:%M:%S. in.txt out.pcap
// The resulting file can be replayed using the WebRTC video_replay tool and
// be inspected in Wireshark using the RTP, VP8 and H264 dissectors.
void SrtpSession::DumpPacket(const void* buf, int len, bool outbound) {
int64_t time_of_day = rtc::TimeUTCMillis() % (24 * 3600 * 1000);
int64_t hours = time_of_day / (3600 * 1000);
int64_t minutes = (time_of_day / (60 * 1000)) % 60;
int64_t seconds = (time_of_day / 1000) % 60;
int64_t millis = time_of_day % 1000;
RTC_LOG(LS_VERBOSE) << "\n" << (outbound ? "O" : "I") << " "
<< std::setfill('0') << std::setw(2) << hours << ":"
<< std::setfill('0') << std::setw(2) << minutes << ":"
<< std::setfill('0') << std::setw(2) << seconds << "."
<< std::setfill('0') << std::setw(3) << millis << " "
<< "000000 " << rtc::hex_encode_with_delimiter((const char *)buf, len, ' ')
<< " # RTP_DUMP";
}
} // namespace cricket