webrtc_m130/p2p/dtls/dtls_transport.cc
Philipp Hancke 589acd56d0 dtls-stun piggybacking: make it compatible with DTLS 1.3
DTLS 1.3 encrypts more parts of the handshake so we move from
deep packet inspection to looking at the state of DTLS to
decide whether to intercept the packet.

BUG=webrtc:367395350

Change-Id: Idb1eda0437f24002f48381af5d6a167a4a153381
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374501
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43794}
2025-01-24 03:11:19 -08:00

940 lines
32 KiB
C++

/*
* Copyright 2011 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 "p2p/dtls/dtls_transport.h"
#include <algorithm>
#include <cstddef>
#include <cstdint>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/crypto/crypto_options.h"
#include "api/dtls_transport_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_transport_state.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_writable_state.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/packet_transport_internal.h"
#include "p2p/dtls/dtls_transport_internal.h"
#include "p2p/dtls/dtls_utils.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/network/ecn_marking.h"
#include "rtc_base/network/received_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/stream.h"
#include "rtc_base/time_utils.h"
namespace cricket {
// We don't pull the RTP constants from rtputils.h, to avoid a layer violation.
static const size_t kMinRtpPacketLen = 12;
// Maximum number of pending packets in the queue. Packets are read immediately
// after they have been written, so a capacity of "1" is sufficient.
//
// However, this bug seems to indicate that's not the case: crbug.com/1063834
// So, temporarily increasing it to 2 to see if that makes a difference.
static const size_t kMaxPendingPackets = 2;
// Minimum and maximum values for the initial DTLS handshake timeout. We'll pick
// an initial timeout based on ICE RTT estimates, but clamp it to this range.
static const int kMinHandshakeTimeoutMs = 50;
static const int kMaxHandshakeTimeoutMs = 3000;
// This effectively disables the handshake timeout.
static const int kDisabledHandshakeTimeoutMs = 3600 * 1000 * 24;
static bool IsRtpPacket(rtc::ArrayView<const uint8_t> payload) {
const uint8_t* u = payload.data();
return (payload.size() >= kMinRtpPacketLen && (u[0] & 0xC0) == 0x80);
}
StreamInterfaceChannel::StreamInterfaceChannel(
IceTransportInternal* ice_transport)
: ice_transport_(ice_transport),
state_(rtc::SS_OPEN),
packets_(kMaxPendingPackets, kMaxDtlsPacketLen) {}
rtc::StreamResult StreamInterfaceChannel::Read(rtc::ArrayView<uint8_t> buffer,
size_t& read,
int& /* error */) {
RTC_DCHECK_RUN_ON(&callback_sequence_);
if (state_ == rtc::SS_CLOSED)
return rtc::SR_EOS;
if (state_ == rtc::SS_OPENING)
return rtc::SR_BLOCK;
if (!packets_.ReadFront(buffer.data(), buffer.size(), &read)) {
return rtc::SR_BLOCK;
}
return rtc::SR_SUCCESS;
}
rtc::StreamResult StreamInterfaceChannel::Write(
rtc::ArrayView<const uint8_t> data,
size_t& written,
int& /* error */) {
RTC_DCHECK_RUN_ON(&callback_sequence_);
// Always succeeds, since this is an unreliable transport anyway.
// TODO(zhihuang): Should this block if ice_transport_'s temporarily
// unwritable?
rtc::PacketOptions packet_options;
ice_transport_->SendPacket(reinterpret_cast<const char*>(data.data()),
data.size(), packet_options);
written = data.size();
return rtc::SR_SUCCESS;
}
bool StreamInterfaceChannel::OnPacketReceived(const char* data, size_t size) {
RTC_DCHECK_RUN_ON(&callback_sequence_);
if (packets_.size() > 0) {
RTC_LOG(LS_WARNING) << "Packet already in queue.";
}
bool ret = packets_.WriteBack(data, size, NULL);
if (!ret) {
// Somehow we received another packet before the SSLStreamAdapter read the
// previous one out of our temporary buffer. In this case, we'll log an
// error and still signal the read event, hoping that it will read the
// packet currently in packets_.
RTC_LOG(LS_ERROR) << "Failed to write packet to queue.";
}
FireEvent(rtc::SE_READ, 0);
return ret;
}
rtc::StreamState StreamInterfaceChannel::GetState() const {
RTC_DCHECK_RUN_ON(&callback_sequence_);
return state_;
}
void StreamInterfaceChannel::Close() {
RTC_DCHECK_RUN_ON(&callback_sequence_);
packets_.Clear();
state_ = rtc::SS_CLOSED;
}
DtlsTransport::DtlsTransport(IceTransportInternal* ice_transport,
const webrtc::CryptoOptions& crypto_options,
webrtc::RtcEventLog* event_log,
rtc::SSLProtocolVersion max_version)
: component_(ice_transport->component()),
ice_transport_(ice_transport),
downward_(NULL),
srtp_ciphers_(crypto_options.GetSupportedDtlsSrtpCryptoSuites()),
ssl_max_version_(max_version),
event_log_(event_log) {
RTC_DCHECK(ice_transport_);
ConnectToIceTransport();
}
DtlsTransport::~DtlsTransport() {
if (ice_transport_) {
ice_transport_->SetPiggybackDtlsDataCallback(nullptr);
ice_transport_->DeregisterReceivedPacketCallback(this);
}
}
webrtc::DtlsTransportState DtlsTransport::dtls_state() const {
return dtls_state_;
}
const std::string& DtlsTransport::transport_name() const {
return ice_transport_->transport_name();
}
int DtlsTransport::component() const {
return component_;
}
bool DtlsTransport::IsDtlsActive() const {
return dtls_active_;
}
bool DtlsTransport::SetLocalCertificate(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
if (dtls_active_) {
if (certificate == local_certificate_) {
// This may happen during renegotiation.
RTC_LOG(LS_INFO) << ToString() << ": Ignoring identical DTLS identity";
return true;
} else {
RTC_LOG(LS_ERROR) << ToString()
<< ": Can't change DTLS local identity in this state";
return false;
}
}
if (certificate) {
local_certificate_ = certificate;
dtls_active_ = true;
} else {
RTC_LOG(LS_INFO) << ToString()
<< ": NULL DTLS identity supplied. Not doing DTLS";
}
return true;
}
rtc::scoped_refptr<rtc::RTCCertificate> DtlsTransport::GetLocalCertificate()
const {
return local_certificate_;
}
bool DtlsTransport::SetDtlsRole(rtc::SSLRole role) {
if (dtls_) {
RTC_DCHECK(dtls_role_);
if (*dtls_role_ != role) {
RTC_LOG(LS_ERROR)
<< "SSL Role can't be reversed after the session is setup.";
return false;
}
return true;
}
dtls_role_ = role;
return true;
}
bool DtlsTransport::GetDtlsRole(rtc::SSLRole* role) const {
if (!dtls_role_) {
return false;
}
*role = *dtls_role_;
return true;
}
bool DtlsTransport::GetSslCipherSuite(int* cipher) const {
if (dtls_state() != webrtc::DtlsTransportState::kConnected) {
return false;
}
return dtls_->GetSslCipherSuite(cipher);
}
std::optional<absl::string_view> DtlsTransport::GetTlsCipherSuiteName() const {
if (dtls_state() != webrtc::DtlsTransportState::kConnected) {
return std::nullopt;
}
return dtls_->GetTlsCipherSuiteName();
}
webrtc::RTCError DtlsTransport::SetRemoteParameters(
absl::string_view digest_alg,
const uint8_t* digest,
size_t digest_len,
std::optional<rtc::SSLRole> role) {
rtc::Buffer remote_fingerprint_value(digest, digest_len);
bool is_dtls_restart =
dtls_active_ && remote_fingerprint_value_ != remote_fingerprint_value;
// Set SSL role. Role must be set before fingerprint is applied, which
// initiates DTLS setup.
if (role) {
if (is_dtls_restart) {
dtls_role_ = *role;
} else {
if (!SetDtlsRole(*role)) {
return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
"Failed to set SSL role for the transport.");
}
}
}
// Apply remote fingerprint.
if (!SetRemoteFingerprint(digest_alg, digest, digest_len)) {
return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
"Failed to apply remote fingerprint.");
}
return webrtc::RTCError::OK();
}
bool DtlsTransport::SetRemoteFingerprint(absl::string_view digest_alg,
const uint8_t* digest,
size_t digest_len) {
rtc::Buffer remote_fingerprint_value(digest, digest_len);
// Once we have the local certificate, the same remote fingerprint can be set
// multiple times.
if (dtls_active_ && remote_fingerprint_value_ == remote_fingerprint_value &&
!digest_alg.empty()) {
// This may happen during renegotiation.
RTC_LOG(LS_INFO) << ToString()
<< ": Ignoring identical remote DTLS fingerprint";
return true;
}
// If the other side doesn't support DTLS, turn off `dtls_active_`.
// TODO(deadbeef): Remove this. It's dangerous, because it relies on higher
// level code to ensure DTLS is actually used, but there are tests that
// depend on it, for the case where an m= section is rejected. In that case
// SetRemoteFingerprint shouldn't even be called though.
if (digest_alg.empty()) {
RTC_DCHECK(!digest_len);
RTC_LOG(LS_INFO) << ToString() << ": Other side didn't support DTLS.";
dtls_active_ = false;
return true;
}
// Otherwise, we must have a local certificate before setting remote
// fingerprint.
if (!dtls_active_) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Can't set DTLS remote settings in this state.";
return false;
}
// At this point we know we are doing DTLS
bool fingerprint_changing = remote_fingerprint_value_.size() > 0u;
remote_fingerprint_value_ = std::move(remote_fingerprint_value);
remote_fingerprint_algorithm_ = std::string(digest_alg);
if (dtls_ && !fingerprint_changing) {
// This can occur if DTLS is set up before a remote fingerprint is
// received. For instance, if we set up DTLS due to receiving an early
// ClientHello.
rtc::SSLPeerCertificateDigestError err = dtls_->SetPeerCertificateDigest(
remote_fingerprint_algorithm_, remote_fingerprint_value_);
if (err != rtc::SSLPeerCertificateDigestError::NONE) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Couldn't set DTLS certificate digest.";
set_dtls_state(webrtc::DtlsTransportState::kFailed);
// If the error is "verification failed", don't return false, because
// this means the fingerprint was formatted correctly but didn't match
// the certificate from the DTLS handshake. Thus the DTLS state should go
// to "failed", but SetRemoteDescription shouldn't fail.
return err == rtc::SSLPeerCertificateDigestError::VERIFICATION_FAILED;
}
return true;
}
// If the fingerprint is changing, we'll tear down the DTLS association and
// create a new one, resetting our state.
if (dtls_ && fingerprint_changing) {
dtls_.reset(nullptr);
set_dtls_state(webrtc::DtlsTransportState::kNew);
set_writable(false);
}
if (!SetupDtls()) {
set_dtls_state(webrtc::DtlsTransportState::kFailed);
return false;
}
return true;
}
std::unique_ptr<rtc::SSLCertChain> DtlsTransport::GetRemoteSSLCertChain()
const {
if (!dtls_) {
return nullptr;
}
return dtls_->GetPeerSSLCertChain();
}
bool DtlsTransport::ExportSrtpKeyingMaterial(
rtc::ZeroOnFreeBuffer<uint8_t>& keying_material) {
return dtls_ ? dtls_->ExportSrtpKeyingMaterial(keying_material) : false;
}
bool DtlsTransport::SetupDtls() {
RTC_DCHECK(dtls_role_);
{
auto downward = std::make_unique<StreamInterfaceChannel>(ice_transport_);
StreamInterfaceChannel* downward_ptr = downward.get();
dtls_ = rtc::SSLStreamAdapter::Create(
std::move(downward),
[this](rtc::SSLHandshakeError error) { OnDtlsHandshakeError(error); },
ice_transport_->field_trials());
if (!dtls_) {
RTC_LOG(LS_ERROR) << ToString() << ": Failed to create DTLS adapter.";
return false;
}
downward_ = downward_ptr;
}
dtls_->SetIdentity(local_certificate_->identity()->Clone());
dtls_->SetMaxProtocolVersion(ssl_max_version_);
dtls_->SetServerRole(*dtls_role_);
dtls_->SetEventCallback(
[this](int events, int err) { OnDtlsEvent(events, err); });
if (remote_fingerprint_value_.size() &&
dtls_->SetPeerCertificateDigest(remote_fingerprint_algorithm_,
remote_fingerprint_value_) !=
rtc::SSLPeerCertificateDigestError::NONE) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Couldn't set DTLS certificate digest.";
return false;
}
// Set up DTLS-SRTP, if it's been enabled.
if (!srtp_ciphers_.empty()) {
if (!dtls_->SetDtlsSrtpCryptoSuites(srtp_ciphers_)) {
RTC_LOG(LS_ERROR) << ToString() << ": Couldn't set DTLS-SRTP ciphers.";
return false;
}
} else {
RTC_LOG(LS_INFO) << ToString() << ": Not using DTLS-SRTP.";
}
RTC_LOG(LS_INFO) << ToString() << ": DTLS setup complete.";
// If the underlying ice_transport is already writable at this point, we may
// be able to start DTLS right away.
MaybeStartDtls();
return true;
}
bool DtlsTransport::GetSrtpCryptoSuite(int* cipher) const {
if (dtls_state() != webrtc::DtlsTransportState::kConnected) {
return false;
}
return dtls_->GetDtlsSrtpCryptoSuite(cipher);
}
bool DtlsTransport::GetSslVersionBytes(int* version) const {
if (dtls_state() != webrtc::DtlsTransportState::kConnected) {
return false;
}
return dtls_->GetSslVersionBytes(version);
}
uint16_t DtlsTransport::GetSslPeerSignatureAlgorithm() const {
if (dtls_state() != webrtc::DtlsTransportState::kConnected) {
return rtc::kSslSignatureAlgorithmUnknown; // "not applicable"
}
return dtls_->GetPeerSignatureAlgorithm();
}
// Called from upper layers to send a media packet.
int DtlsTransport::SendPacket(const char* data,
size_t size,
const rtc::PacketOptions& options,
int flags) {
if (!dtls_active_) {
// Not doing DTLS.
return ice_transport_->SendPacket(data, size, options);
}
switch (dtls_state()) {
case webrtc::DtlsTransportState::kNew:
// Can't send data until the connection is active.
// TODO(ekr@rtfm.com): assert here if dtls_ is NULL?
return -1;
case webrtc::DtlsTransportState::kConnecting:
// Can't send data until the connection is active.
return -1;
case webrtc::DtlsTransportState::kConnected:
if (flags & PF_SRTP_BYPASS) {
RTC_DCHECK(!srtp_ciphers_.empty());
if (!IsRtpPacket(rtc::MakeArrayView(
reinterpret_cast<const uint8_t*>(data), size))) {
return -1;
}
return ice_transport_->SendPacket(data, size, options);
} else {
size_t written;
int error;
return (dtls_->WriteAll(
rtc::MakeArrayView(reinterpret_cast<const uint8_t*>(data),
size),
written, error) == rtc::SR_SUCCESS)
? static_cast<int>(size)
: -1;
}
case webrtc::DtlsTransportState::kFailed:
// Can't send anything when we're failed.
RTC_LOG(LS_ERROR) << ToString()
<< ": Couldn't send packet due to "
"webrtc::DtlsTransportState::kFailed.";
return -1;
case webrtc::DtlsTransportState::kClosed:
// Can't send anything when we're closed.
RTC_LOG(LS_ERROR) << ToString()
<< ": Couldn't send packet due to "
"webrtc::DtlsTransportState::kClosed.";
return -1;
default:
RTC_DCHECK_NOTREACHED();
return -1;
}
}
IceTransportInternal* DtlsTransport::ice_transport() {
return ice_transport_;
}
bool DtlsTransport::IsDtlsConnected() {
return dtls_ && dtls_->IsTlsConnected();
}
bool DtlsTransport::receiving() const {
return receiving_;
}
bool DtlsTransport::writable() const {
return writable_;
}
int DtlsTransport::GetError() {
return ice_transport_->GetError();
}
std::optional<rtc::NetworkRoute> DtlsTransport::network_route() const {
return ice_transport_->network_route();
}
bool DtlsTransport::GetOption(rtc::Socket::Option opt, int* value) {
return ice_transport_->GetOption(opt, value);
}
int DtlsTransport::SetOption(rtc::Socket::Option opt, int value) {
return ice_transport_->SetOption(opt, value);
}
void DtlsTransport::ConnectToIceTransport() {
RTC_DCHECK(ice_transport_);
ice_transport_->SignalWritableState.connect(this,
&DtlsTransport::OnWritableState);
ice_transport_->RegisterReceivedPacketCallback(
this, [&](rtc::PacketTransportInternal* transport,
const rtc::ReceivedPacket& packet) {
OnReadPacket(transport, packet);
});
ice_transport_->SignalSentPacket.connect(this, &DtlsTransport::OnSentPacket);
ice_transport_->SignalReadyToSend.connect(this,
&DtlsTransport::OnReadyToSend);
ice_transport_->SignalReceivingState.connect(
this, &DtlsTransport::OnReceivingState);
ice_transport_->SignalNetworkRouteChanged.connect(
this, &DtlsTransport::OnNetworkRouteChanged);
ice_transport_->SetPiggybackDtlsDataCallback(
[this](rtc::PacketTransportInternal* transport,
const rtc::ReceivedPacket& packet) {
RTC_DCHECK(dtls_active_);
RTC_DCHECK(IsDtlsPacket(packet.payload()));
if (!dtls_active_) {
// Not doing DTLS.
return;
}
if (!IsDtlsPacket(packet.payload())) {
return;
}
OnReadPacket(transport, packet);
});
}
// The state transition logic here is as follows:
// (1) If we're not doing DTLS-SRTP, then the state is just the
// state of the underlying impl()
// (2) If we're doing DTLS-SRTP:
// - Prior to the DTLS handshake, the state is neither receiving nor
// writable
// - When the impl goes writable for the first time we
// start the DTLS handshake
// - Once the DTLS handshake completes, the state is that of the
// impl again
void DtlsTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(transport == ice_transport_);
RTC_LOG(LS_VERBOSE) << ToString()
<< ": ice_transport writable state changed to "
<< ice_transport_->writable();
if (!dtls_active_) {
// Not doing DTLS.
// Note: SignalWritableState fired by set_writable.
set_writable(ice_transport_->writable());
return;
}
// The opportunistic attempt to do DTLS piggybacking failed.
// Recreate the DTLS session. Note: this assumes we can consider
// the previous DTLS session state beyond repair and no packet
// reached the peer.
if (ice_transport_->config().dtls_handshake_in_stun && dtls_ &&
!was_ever_connected_ &&
!ice_transport_->IsDtlsPiggybackSupportedByPeer() &&
(dtls_state() == webrtc::DtlsTransportState::kConnecting ||
dtls_state() == webrtc::DtlsTransportState::kNew)) {
RTC_LOG(LS_INFO) << "DTLS piggybacking not supported, restarting...";
ice_transport_->SetPiggybackDtlsDataCallback(nullptr);
dtls_.reset(nullptr);
set_dtls_state(webrtc::DtlsTransportState::kNew);
set_writable(false);
if (!SetupDtls()) {
RTC_LOG(LS_ERROR)
<< "Failed to setup DTLS again after attempted piggybacking.";
set_dtls_state(webrtc::DtlsTransportState::kFailed);
return;
}
// SetupDtls has called MaybeStartDtls() already.
return;
}
switch (dtls_state()) {
case webrtc::DtlsTransportState::kNew:
MaybeStartDtls();
break;
case webrtc::DtlsTransportState::kConnected:
was_ever_connected_ = true;
// Note: SignalWritableState fired by set_writable.
set_writable(ice_transport_->writable());
break;
case webrtc::DtlsTransportState::kConnecting:
// Do nothing.
break;
case webrtc::DtlsTransportState::kFailed:
// Should not happen. Do nothing.
RTC_LOG(LS_ERROR) << ToString()
<< ": OnWritableState() called in state "
"webrtc::DtlsTransportState::kFailed.";
break;
case webrtc::DtlsTransportState::kClosed:
// Should not happen. Do nothing.
RTC_LOG(LS_ERROR) << ToString()
<< ": OnWritableState() called in state "
"webrtc::DtlsTransportState::kClosed.";
break;
case webrtc::DtlsTransportState::kNumValues:
RTC_DCHECK_NOTREACHED();
break;
}
}
void DtlsTransport::OnReceivingState(rtc::PacketTransportInternal* transport) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(transport == ice_transport_);
RTC_LOG(LS_VERBOSE) << ToString()
<< ": ice_transport "
"receiving state changed to "
<< ice_transport_->receiving();
if (!dtls_active_ || dtls_state() == webrtc::DtlsTransportState::kConnected) {
// Note: SignalReceivingState fired by set_receiving.
set_receiving(ice_transport_->receiving());
}
}
void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
const rtc::ReceivedPacket& packet) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(transport == ice_transport_);
if (!dtls_active_) {
// Not doing DTLS.
NotifyPacketReceived(packet);
return;
}
switch (dtls_state()) {
case webrtc::DtlsTransportState::kNew:
if (dtls_) {
RTC_LOG(LS_INFO) << ToString()
<< ": Packet received before DTLS started.";
} else {
RTC_LOG(LS_WARNING) << ToString()
<< ": Packet received before we know if we are "
"doing DTLS or not.";
}
// Cache a client hello packet received before DTLS has actually started.
if (IsDtlsClientHelloPacket(packet.payload())) {
RTC_LOG(LS_INFO) << ToString()
<< ": Caching DTLS ClientHello packet until DTLS is "
"started.";
cached_client_hello_.SetData(packet.payload());
// If we haven't started setting up DTLS yet (because we don't have a
// remote fingerprint/role), we can use the client hello as a clue that
// the peer has chosen the client role, and proceed with the handshake.
// The fingerprint will be verified when it's set.
if (!dtls_ && local_certificate_) {
SetDtlsRole(rtc::SSL_SERVER);
SetupDtls();
}
} else {
RTC_LOG(LS_INFO) << ToString()
<< ": Not a DTLS ClientHello packet; dropping.";
}
break;
case webrtc::DtlsTransportState::kConnecting:
case webrtc::DtlsTransportState::kConnected:
// We should only get DTLS or SRTP packets; STUN's already been demuxed.
// Is this potentially a DTLS packet?
if (IsDtlsPacket(packet.payload())) {
if (!HandleDtlsPacket(packet.payload())) {
RTC_LOG(LS_ERROR) << ToString() << ": Failed to handle DTLS packet.";
return;
}
} else {
// Not a DTLS packet; our handshake should be complete by now.
if (dtls_state() != webrtc::DtlsTransportState::kConnected) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Received non-DTLS packet before DTLS "
"complete.";
return;
}
// And it had better be a SRTP packet.
if (!IsRtpPacket(packet.payload())) {
RTC_LOG(LS_ERROR)
<< ToString() << ": Received unexpected non-DTLS packet.";
return;
}
// Sanity check.
RTC_DCHECK(!srtp_ciphers_.empty());
// Signal this upwards as a bypass packet.
NotifyPacketReceived(
packet.CopyAndSet(rtc::ReceivedPacket::kSrtpEncrypted));
}
break;
case webrtc::DtlsTransportState::kFailed:
case webrtc::DtlsTransportState::kClosed:
case webrtc::DtlsTransportState::kNumValues:
// This shouldn't be happening. Drop the packet.
break;
}
}
void DtlsTransport::OnSentPacket(rtc::PacketTransportInternal* /* transport */,
const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(&thread_checker_);
SignalSentPacket(this, sent_packet);
}
void DtlsTransport::OnReadyToSend(
rtc::PacketTransportInternal* /* transport */) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (writable()) {
SignalReadyToSend(this);
}
}
void DtlsTransport::OnDtlsEvent(int sig, int err) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(dtls_);
if (sig & rtc::SE_OPEN) {
// This is the first time.
RTC_LOG(LS_INFO) << ToString() << ": DTLS handshake complete.";
// The check for OPEN shouldn't be necessary but let's make
// sure we don't accidentally frob the state if it's closed.
if (dtls_->GetState() == rtc::SS_OPEN) {
int ssl_version_bytes;
bool ret = dtls_->GetSslVersionBytes(&ssl_version_bytes);
RTC_DCHECK(ret);
ice_transport_->SetDtlsHandshakeComplete(
dtls_role_ == rtc::SSL_CLIENT,
ssl_version_bytes == rtc::kDtls13VersionBytes);
set_dtls_state(webrtc::DtlsTransportState::kConnected);
set_writable(true);
}
}
if (sig & rtc::SE_READ) {
uint8_t buf[kMaxDtlsPacketLen];
size_t read;
int read_error;
rtc::StreamResult ret;
// The underlying DTLS stream may have received multiple DTLS records in
// one packet, so read all of them.
do {
ret = dtls_->Read(buf, read, read_error);
if (ret == rtc::SR_SUCCESS) {
// TODO(bugs.webrtc.org/15368): It should be possible to use information
// from the original packet here to populate socket address and
// timestamp.
NotifyPacketReceived(rtc::ReceivedPacket(
rtc::MakeArrayView(buf, read), rtc::SocketAddress(),
webrtc::Timestamp::Micros(rtc::TimeMicros()),
rtc::EcnMarking::kNotEct, rtc::ReceivedPacket::kDtlsDecrypted));
} else if (ret == rtc::SR_EOS) {
// Remote peer shut down the association with no error.
RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed by remote";
set_writable(false);
set_dtls_state(webrtc::DtlsTransportState::kClosed);
NotifyOnClose();
} else if (ret == rtc::SR_ERROR) {
// Remote peer shut down the association with an error.
RTC_LOG(LS_INFO)
<< ToString()
<< ": Closed by remote with DTLS transport error, code="
<< read_error;
set_writable(false);
set_dtls_state(webrtc::DtlsTransportState::kFailed);
NotifyOnClose();
}
} while (ret == rtc::SR_SUCCESS);
}
if (sig & rtc::SE_CLOSE) {
RTC_DCHECK(sig == rtc::SE_CLOSE); // SE_CLOSE should be by itself.
set_writable(false);
if (!err) {
RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed";
set_dtls_state(webrtc::DtlsTransportState::kClosed);
} else {
RTC_LOG(LS_INFO) << ToString() << ": DTLS transport error, code=" << err;
set_dtls_state(webrtc::DtlsTransportState::kFailed);
}
}
}
void DtlsTransport::OnNetworkRouteChanged(
std::optional<rtc::NetworkRoute> network_route) {
RTC_DCHECK_RUN_ON(&thread_checker_);
SignalNetworkRouteChanged(network_route);
}
void DtlsTransport::MaybeStartDtls() {
RTC_DCHECK(ice_transport_);
// When adding the DTLS handshake in STUN we want to call StartSSL even
// before the ICE transport is ready.
bool start_early_for_dtls_in_stun =
ice_transport_->config().dtls_handshake_in_stun;
if (dtls_ && (ice_transport_->writable() || start_early_for_dtls_in_stun)) {
ConfigureHandshakeTimeout(start_early_for_dtls_in_stun);
if (dtls_->StartSSL()) {
// This should never fail:
// Because we are operating in a nonblocking mode and all
// incoming packets come in via OnReadPacket(), which rejects
// packets in this state, the incoming queue must be empty. We
// ignore write errors, thus any errors must be because of
// configuration and therefore are our fault.
RTC_DCHECK_NOTREACHED() << "StartSSL failed.";
RTC_LOG(LS_ERROR) << ToString() << ": Couldn't start DTLS handshake";
set_dtls_state(webrtc::DtlsTransportState::kFailed);
return;
}
RTC_LOG(LS_INFO) << ToString()
<< ": DtlsTransport: Started DTLS handshake active="
<< IsDtlsActive();
set_dtls_state(webrtc::DtlsTransportState::kConnecting);
// Now that the handshake has started, we can process a cached ClientHello
// (if one exists).
if (cached_client_hello_.size()) {
if (*dtls_role_ == rtc::SSL_SERVER) {
RTC_LOG(LS_INFO) << ToString()
<< ": Handling cached DTLS ClientHello packet.";
if (!HandleDtlsPacket(cached_client_hello_)) {
RTC_LOG(LS_ERROR) << ToString() << ": Failed to handle DTLS packet.";
}
} else {
RTC_LOG(LS_WARNING) << ToString()
<< ": Discarding cached DTLS ClientHello packet "
"because we don't have the server role.";
}
cached_client_hello_.Clear();
}
}
}
// Called from OnReadPacket when a DTLS packet is received.
bool DtlsTransport::HandleDtlsPacket(rtc::ArrayView<const uint8_t> payload) {
// Pass to the StreamInterfaceChannel which ends up being passed to the DTLS
// stack.
return downward_->OnPacketReceived(
reinterpret_cast<const char*>(payload.data()), payload.size());
}
void DtlsTransport::set_receiving(bool receiving) {
if (receiving_ == receiving) {
return;
}
receiving_ = receiving;
SignalReceivingState(this);
}
void DtlsTransport::set_writable(bool writable) {
if (writable_ == writable) {
return;
}
if (event_log_) {
event_log_->Log(
std::make_unique<webrtc::RtcEventDtlsWritableState>(writable));
}
RTC_LOG(LS_VERBOSE) << ToString() << ": set_writable to: " << writable;
writable_ = writable;
if (writable_) {
SignalReadyToSend(this);
}
SignalWritableState(this);
}
void DtlsTransport::set_dtls_state(webrtc::DtlsTransportState state) {
if (dtls_state_ == state) {
return;
}
if (event_log_) {
event_log_->Log(
std::make_unique<webrtc::RtcEventDtlsTransportState>(state));
}
RTC_LOG(LS_VERBOSE) << ToString() << ": set_dtls_state from:"
<< static_cast<int>(dtls_state_) << " to "
<< static_cast<int>(state);
dtls_state_ = state;
SendDtlsState(this, state);
}
void DtlsTransport::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
SendDtlsHandshakeError(error);
}
void DtlsTransport::ConfigureHandshakeTimeout(bool uses_dtls_in_stun) {
RTC_DCHECK(dtls_);
std::optional<int> rtt_ms = ice_transport_->GetRttEstimate();
if (uses_dtls_in_stun) {
// Configure a very high timeout to effectively disable the DTLS timeout
// and avoid fragmented resends. This is ok since DTLS-in-STUN caches
// the handshake pacets and resends them using the pacing of ICE.
RTC_LOG(LS_INFO) << ToString() << ": configuring DTLS handshake timeout "
<< kDisabledHandshakeTimeoutMs << "ms for DTLS-in-STUN";
dtls_->SetInitialRetransmissionTimeout(kDisabledHandshakeTimeoutMs);
} else if (rtt_ms) {
// Limit the timeout to a reasonable range in case the ICE RTT takes
// extreme values.
int initial_timeout_ms =
std::max(kMinHandshakeTimeoutMs,
std::min(kMaxHandshakeTimeoutMs, 2 * (*rtt_ms)));
RTC_LOG(LS_INFO) << ToString() << ": configuring DTLS handshake timeout "
<< initial_timeout_ms << "ms based on ICE RTT " << *rtt_ms;
dtls_->SetInitialRetransmissionTimeout(initial_timeout_ms);
} else {
RTC_LOG(LS_INFO)
<< ToString()
<< ": no RTT estimate - using default DTLS handshake timeout";
}
}
} // namespace cricket