| /* |
| * 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/base/dtls_transport.h" |
| |
| #include <algorithm> |
| #include <cstdint> |
| #include <memory> |
| #include <utility> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/strings/string_view.h" |
| #include "api/array_view.h" |
| #include "api/dtls_transport_interface.h" |
| #include "api/rtc_event_log/rtc_event_log.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/packet_transport_internal.h" |
| #include "rtc_base/buffer.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/dscp.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/network/received_packet.h" |
| #include "rtc_base/rtc_certificate.h" |
| #include "rtc_base/socket_address.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/stream.h" |
| #include "rtc_base/thread.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 kDtlsRecordHeaderLen = 13; |
| static const size_t kMaxDtlsPacketLen = 2048; |
| 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 kMinHandshakeTimeout = 50; |
| static const int kMaxHandshakeTimeout = 3000; |
| |
| static bool IsDtlsPacket(rtc::ArrayView<const uint8_t> payload) { |
| const uint8_t* u = payload.data(); |
| return (payload.size() >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64)); |
| } |
| static bool IsDtlsClientHelloPacket(rtc::ArrayView<const uint8_t> payload) { |
| if (!IsDtlsPacket(payload)) { |
| return false; |
| } |
| const uint8_t* u = payload.data(); |
| return payload.size() > 17 && u[0] == 22 && u[13] == 1; |
| } |
| 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_->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) { |
| if (dtls_state() != webrtc::DtlsTransportState::kConnected) { |
| return false; |
| } |
| |
| return dtls_->GetSslCipherSuite(cipher); |
| } |
| |
| webrtc::RTCError DtlsTransport::SetRemoteParameters( |
| absl::string_view digest_alg, |
| const uint8_t* digest, |
| size_t digest_len, |
| absl::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; |
| if (!dtls_->SetPeerCertificateDigest( |
| remote_fingerprint_algorithm_, |
| reinterpret_cast<unsigned char*>(remote_fingerprint_value_.data()), |
| remote_fingerprint_value_.size(), &err)) { |
| 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::ExportKeyingMaterial(absl::string_view label, |
| const uint8_t* context, |
| size_t context_len, |
| bool use_context, |
| uint8_t* result, |
| size_t result_len) { |
| return (dtls_.get()) |
| ? dtls_->ExportKeyingMaterial(label, context, context_len, |
| use_context, result, result_len) |
| : 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); }); |
| 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_, |
| reinterpret_cast<unsigned char*>(remote_fingerprint_value_.data()), |
| remote_fingerprint_value_.size())) { |
| 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) { |
| 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(); |
| } |
| |
| absl::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); |
| } |
| |
| // 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; |
| } |
| |
| switch (dtls_state()) { |
| case webrtc::DtlsTransportState::kNew: |
| MaybeStartDtls(); |
| break; |
| case webrtc::DtlsTransportState::kConnected: |
| // 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_); |
| if (sig & rtc::SE_OPEN) { |
| // This is the first time. |
| RTC_LOG(LS_INFO) << ToString() << ": DTLS handshake complete."; |
| if (dtls_->GetState() == rtc::SS_OPEN) { |
| // The check for OPEN shouldn't be necessary but let's make |
| // sure we don't accidentally frob the state if it's closed. |
| 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( |
| absl::optional<rtc::NetworkRoute> network_route) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| SignalNetworkRouteChanged(network_route); |
| } |
| |
| void DtlsTransport::MaybeStartDtls() { |
| if (dtls_ && ice_transport_->writable()) { |
| ConfigureHandshakeTimeout(); |
| |
| 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) { |
| // Sanity check we're not passing junk that |
| // just looks like DTLS. |
| const uint8_t* tmp_data = payload.data(); |
| size_t tmp_size = payload.size(); |
| while (tmp_size > 0) { |
| if (tmp_size < kDtlsRecordHeaderLen) |
| return false; // Too short for the header |
| |
| size_t record_len = (tmp_data[11] << 8) | (tmp_data[12]); |
| if ((record_len + kDtlsRecordHeaderLen) > tmp_size) |
| return false; // Body too short |
| |
| tmp_data += record_len + kDtlsRecordHeaderLen; |
| tmp_size -= record_len + kDtlsRecordHeaderLen; |
| } |
| |
| // Looks good. Pass to the SIC 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() { |
| RTC_DCHECK(dtls_); |
| absl::optional<int> rtt = ice_transport_->GetRttEstimate(); |
| if (rtt) { |
| // Limit the timeout to a reasonable range in case the ICE RTT takes |
| // extreme values. |
| int initial_timeout = std::max(kMinHandshakeTimeout, |
| std::min(kMaxHandshakeTimeout, 2 * (*rtt))); |
| RTC_LOG(LS_INFO) << ToString() << ": configuring DTLS handshake timeout " |
| << initial_timeout << " based on ICE RTT " << *rtt; |
| |
| dtls_->SetInitialRetransmissionTimeout(initial_timeout); |
| } else { |
| RTC_LOG(LS_INFO) |
| << ToString() |
| << ": no RTT estimate - using default DTLS handshake timeout"; |
| } |
| } |
| |
| } // namespace cricket |