blob: bf8583f670a8a51e33c97ab54baa259137f063eb [file] [log] [blame]
/*
* 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 <memory>
#include <utility>
#include "absl/memory/memory.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/rtc_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/stream.h"
#include "rtc_base/thread.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(const char* data, size_t len) {
const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
return (len >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64));
}
static bool IsDtlsClientHelloPacket(const char* data, size_t len) {
if (!IsDtlsPacket(data, len)) {
return false;
}
const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
return len > 17 && u[0] == 22 && u[13] == 1;
}
static bool IsRtpPacket(const char* data, size_t len) {
const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
return (len >= 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(void* buffer,
size_t buffer_len,
size_t* read,
int* error) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (state_ == rtc::SS_CLOSED)
return rtc::SR_EOS;
if (state_ == rtc::SS_OPENING)
return rtc::SR_BLOCK;
if (!packets_.ReadFront(buffer, buffer_len, read)) {
return rtc::SR_BLOCK;
}
return rtc::SR_SUCCESS;
}
rtc::StreamResult StreamInterfaceChannel::Write(const void* data,
size_t data_len,
size_t* written,
int* error) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// 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(static_cast<const char*>(data), data_len,
packet_options);
if (written) {
*written = data_len;
}
return rtc::SR_SUCCESS;
}
bool StreamInterfaceChannel::OnPacketReceived(const char* data, size_t size) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
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.";
}
SignalEvent(this, rtc::SE_READ, 0);
return ret;
}
rtc::StreamState StreamInterfaceChannel::GetState() const {
RTC_DCHECK_RUN_ON(&sequence_checker_);
return state_;
}
void StreamInterfaceChannel::Close() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
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() = default;
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);
}
bool DtlsTransport::SetRemoteFingerprint(const std::string& 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_ = 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(const std::string& 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));
if (!dtls_) {
RTC_LOG(LS_ERROR) << ToString() << ": Failed to create DTLS adapter.";
return false;
}
downward_ = downward_ptr;
}
dtls_->SetIdentity(local_certificate_->identity()->Clone());
dtls_->SetMode(rtc::SSL_MODE_DTLS);
dtls_->SetMaxProtocolVersion(ssl_max_version_);
dtls_->SetServerRole(*dtls_role_);
dtls_->SignalEvent.connect(this, &DtlsTransport::OnDtlsEvent);
dtls_->SignalSSLHandshakeError.connect(this,
&DtlsTransport::OnDtlsHandshakeError);
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);
}
// 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(data, size)) {
return -1;
}
return ice_transport_->SendPacket(data, size, options);
} else {
return (dtls_->WriteAll(data, size, NULL, NULL) == 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_->SignalReadPacket.connect(this, &DtlsTransport::OnReadPacket);
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 char* data,
size_t size,
const int64_t& packet_time_us,
int flags) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(transport == ice_transport_);
RTC_DCHECK(flags == 0);
if (!dtls_active_) {
// Not doing DTLS.
SignalReadPacket(this, data, size, packet_time_us, 0);
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(data, size)) {
RTC_LOG(LS_INFO) << ToString()
<< ": Caching DTLS ClientHello packet until DTLS is "
"started.";
cached_client_hello_.SetData(data, size);
// 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(data, size)) {
if (!HandleDtlsPacket(data, size)) {
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(data, size)) {
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.
SignalReadPacket(this, data, size, packet_time_us, PF_SRTP_BYPASS);
}
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(rtc::StreamInterface* dtls, int sig, int err) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(dtls == dtls_.get());
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) {
char 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, sizeof(buf), &read, &read_error);
if (ret == rtc::SR_SUCCESS) {
SignalReadPacket(this, buf, read, rtc::TimeMicros(), 0);
} 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);
SignalClosed(this);
} 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);
SignalClosed(this);
}
} 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";
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_.data<char>(),
cached_client_hello_.size())) {
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(const char* data, size_t size) {
// Sanity check we're not passing junk that
// just looks like DTLS.
const uint8_t* tmp_data = reinterpret_cast<const uint8_t*>(data);
size_t tmp_size = 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(data, 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