mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00

This is a follow-up to: https://webrtc-review.googlesource.com/c/src/+/318640 The problem was that the scoped field trials in the tests only applied to the construction of the streams, not the handshake. Note, although the changes are in OpenSSLStreamAdapter, this CL actually fixes the SSLStreamAdapterTestDTLSExtensionPermutation tests in rtc_base/ssl_stream_adapter_unittest.cc. Bug: webrtc:15467 Change-Id: I25cdd758aab1bc67fd7a6a61c956c6d52f82e3d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344762 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41976}
259 lines
9.3 KiB
C++
259 lines
9.3 KiB
C++
/*
|
|
* Copyright 2004 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.
|
|
*/
|
|
|
|
#ifndef RTC_BASE_OPENSSL_STREAM_ADAPTER_H_
|
|
#define RTC_BASE_OPENSSL_STREAM_ADAPTER_H_
|
|
|
|
#include <openssl/ossl_typ.h>
|
|
#include <stddef.h>
|
|
#include <stdint.h>
|
|
|
|
#include <memory>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "absl/functional/any_invocable.h"
|
|
#include "absl/strings/string_view.h"
|
|
#include "absl/types/optional.h"
|
|
#include "rtc_base/buffer.h"
|
|
#ifdef OPENSSL_IS_BORINGSSL
|
|
#include "rtc_base/boringssl_identity.h"
|
|
#else
|
|
#include "rtc_base/openssl_identity.h"
|
|
#endif
|
|
#include "api/task_queue/pending_task_safety_flag.h"
|
|
#include "rtc_base/ssl_identity.h"
|
|
#include "rtc_base/ssl_stream_adapter.h"
|
|
#include "rtc_base/stream.h"
|
|
#include "rtc_base/system/rtc_export.h"
|
|
#include "rtc_base/task_utils/repeating_task.h"
|
|
#include "rtc_base/third_party/sigslot/sigslot.h"
|
|
|
|
namespace rtc {
|
|
|
|
// This class was written with OpenSSLAdapter (a socket adapter) as a
|
|
// starting point. It has similar structure and functionality, but uses a
|
|
// "peer-to-peer" mode, verifying the peer's certificate using a digest
|
|
// sent over a secure signaling channel.
|
|
//
|
|
// Static methods to initialize and deinit the SSL library are in
|
|
// OpenSSLAdapter. These should probably be moved out to a neutral class.
|
|
//
|
|
// In a few cases I have factored out some OpenSSLAdapter code into static
|
|
// methods so it can be reused from this class. Eventually that code should
|
|
// probably be moved to a common support class. Unfortunately there remain a
|
|
// few duplicated sections of code. I have not done more restructuring because
|
|
// I did not want to affect existing code that uses OpenSSLAdapter.
|
|
//
|
|
// This class does not support the SSL connection restart feature present in
|
|
// OpenSSLAdapter. I am not entirely sure how the feature is useful and I am
|
|
// not convinced that it works properly.
|
|
//
|
|
// This implementation is careful to disallow data exchange after an SSL error,
|
|
// and it has an explicit SSL_CLOSED state. It should not be possible to send
|
|
// any data in clear after one of the StartSSL methods has been called.
|
|
|
|
// Look in sslstreamadapter.h for documentation of the methods.
|
|
|
|
class SSLCertChain;
|
|
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
|
|
class OpenSSLStreamAdapter final : public SSLStreamAdapter,
|
|
public sigslot::has_slots<> {
|
|
public:
|
|
OpenSSLStreamAdapter(
|
|
std::unique_ptr<StreamInterface> stream,
|
|
absl::AnyInvocable<void(SSLHandshakeError)> handshake_error);
|
|
~OpenSSLStreamAdapter() override;
|
|
|
|
void SetIdentity(std::unique_ptr<SSLIdentity> identity) override;
|
|
SSLIdentity* GetIdentityForTesting() const override;
|
|
|
|
// Default argument is for compatibility
|
|
void SetServerRole(SSLRole role = SSL_SERVER) override;
|
|
bool SetPeerCertificateDigest(
|
|
absl::string_view digest_alg,
|
|
const unsigned char* digest_val,
|
|
size_t digest_len,
|
|
SSLPeerCertificateDigestError* error = nullptr) override;
|
|
|
|
std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const override;
|
|
|
|
// Goes from state SSL_NONE to either SSL_CONNECTING or SSL_WAIT, depending
|
|
// on whether the underlying stream is already open or not.
|
|
int StartSSL() override;
|
|
void SetMode(SSLMode mode) override;
|
|
void SetMaxProtocolVersion(SSLProtocolVersion version) override;
|
|
void SetInitialRetransmissionTimeout(int timeout_ms) override;
|
|
|
|
StreamResult Read(rtc::ArrayView<uint8_t> data,
|
|
size_t& read,
|
|
int& error) override;
|
|
StreamResult Write(rtc::ArrayView<const uint8_t> data,
|
|
size_t& written,
|
|
int& error) override;
|
|
void Close() override;
|
|
StreamState GetState() const override;
|
|
|
|
// TODO(guoweis): Move this away from a static class method.
|
|
static std::string SslCipherSuiteToName(int crypto_suite);
|
|
|
|
bool GetSslCipherSuite(int* cipher) override;
|
|
|
|
SSLProtocolVersion GetSslVersion() const override;
|
|
bool GetSslVersionBytes(int* version) const override;
|
|
// Key Extractor interface
|
|
bool ExportKeyingMaterial(absl::string_view label,
|
|
const uint8_t* context,
|
|
size_t context_len,
|
|
bool use_context,
|
|
uint8_t* result,
|
|
size_t result_len) override;
|
|
|
|
uint16_t GetPeerSignatureAlgorithm() const override;
|
|
|
|
// DTLS-SRTP interface
|
|
bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites) override;
|
|
bool GetDtlsSrtpCryptoSuite(int* crypto_suite) override;
|
|
|
|
bool IsTlsConnected() override;
|
|
|
|
// Capabilities interfaces.
|
|
static bool IsBoringSsl();
|
|
|
|
static bool IsAcceptableCipher(int cipher, KeyType key_type);
|
|
static bool IsAcceptableCipher(absl::string_view cipher, KeyType key_type);
|
|
|
|
// Use our timeutils.h source of timing in BoringSSL, allowing us to test
|
|
// using a fake clock.
|
|
static void EnableTimeCallbackForTesting();
|
|
|
|
private:
|
|
enum SSLState {
|
|
// Before calling one of the StartSSL methods, data flows
|
|
// in clear text.
|
|
SSL_NONE,
|
|
SSL_WAIT, // waiting for the stream to open to start SSL negotiation
|
|
SSL_CONNECTING, // SSL negotiation in progress
|
|
SSL_CONNECTED, // SSL stream successfully established
|
|
SSL_ERROR, // some SSL error occurred, stream is closed
|
|
SSL_CLOSED // Clean close
|
|
};
|
|
|
|
void OnEvent(StreamInterface* stream, int events, int err);
|
|
|
|
void PostEvent(int events, int err);
|
|
void SetTimeout(int delay_ms);
|
|
|
|
// The following three methods return 0 on success and a negative
|
|
// error code on failure. The error code may be from OpenSSL or -1
|
|
// on some other error cases, so it can't really be interpreted
|
|
// unfortunately.
|
|
|
|
// Prepare SSL library, state is SSL_CONNECTING.
|
|
int BeginSSL();
|
|
// Perform SSL negotiation steps.
|
|
int ContinueSSL();
|
|
|
|
// Error handler helper. signal is given as true for errors in
|
|
// asynchronous contexts (when an error method was not returned
|
|
// through some other method), and in that case an SE_CLOSE event is
|
|
// raised on the stream with the specified error.
|
|
// A 0 error means a graceful close, otherwise there is not really enough
|
|
// context to interpret the error code.
|
|
// `alert` indicates an alert description (one of the SSL_AD constants) to
|
|
// send to the remote endpoint when closing the association. If 0, a normal
|
|
// shutdown will be performed.
|
|
void Error(absl::string_view context, int err, uint8_t alert, bool signal);
|
|
void Cleanup(uint8_t alert);
|
|
|
|
// Flush the input buffers by reading left bytes (for DTLS)
|
|
void FlushInput(unsigned int left);
|
|
|
|
// SSL library configuration
|
|
SSL_CTX* SetupSSLContext();
|
|
// Verify the peer certificate matches the signaled digest.
|
|
bool VerifyPeerCertificate();
|
|
|
|
#ifdef OPENSSL_IS_BORINGSSL
|
|
// SSL certificate verification callback. See SSL_CTX_set_custom_verify.
|
|
static enum ssl_verify_result_t SSLVerifyCallback(SSL* ssl,
|
|
uint8_t* out_alert);
|
|
#else
|
|
// SSL certificate verification callback. See
|
|
// SSL_CTX_set_cert_verify_callback.
|
|
static int SSLVerifyCallback(X509_STORE_CTX* store, void* arg);
|
|
#endif
|
|
|
|
bool WaitingToVerifyPeerCertificate() const {
|
|
return GetClientAuthEnabled() && !peer_certificate_verified_;
|
|
}
|
|
|
|
bool HasPeerCertificateDigest() const {
|
|
return !peer_certificate_digest_algorithm_.empty() &&
|
|
!peer_certificate_digest_value_.empty();
|
|
}
|
|
|
|
const std::unique_ptr<StreamInterface> stream_;
|
|
absl::AnyInvocable<void(SSLHandshakeError)> handshake_error_;
|
|
|
|
rtc::Thread* const owner_;
|
|
webrtc::ScopedTaskSafety task_safety_;
|
|
webrtc::RepeatingTaskHandle timeout_task_;
|
|
|
|
SSLState state_;
|
|
SSLRole role_;
|
|
int ssl_error_code_; // valid when state_ == SSL_ERROR or SSL_CLOSED
|
|
// Whether the SSL negotiation is blocked on needing to read or
|
|
// write to the wrapped stream.
|
|
bool ssl_read_needs_write_;
|
|
bool ssl_write_needs_read_;
|
|
|
|
SSL* ssl_;
|
|
SSL_CTX* ssl_ctx_;
|
|
|
|
// Our key and certificate.
|
|
#ifdef OPENSSL_IS_BORINGSSL
|
|
std::unique_ptr<BoringSSLIdentity> identity_;
|
|
// We check and store the `WebRTC-PermuteTlsClientHello` field trial config in
|
|
// the constructor for convenience to allow tests to apply different
|
|
// configurations across instances.
|
|
const bool permute_extension_;
|
|
#else
|
|
std::unique_ptr<OpenSSLIdentity> identity_;
|
|
#endif
|
|
// The certificate chain that the peer presented. Initially null, until the
|
|
// connection is established.
|
|
std::unique_ptr<SSLCertChain> peer_cert_chain_;
|
|
bool peer_certificate_verified_ = false;
|
|
// The digest of the certificate that the peer must present.
|
|
Buffer peer_certificate_digest_value_;
|
|
std::string peer_certificate_digest_algorithm_;
|
|
|
|
// The DtlsSrtp ciphers
|
|
std::string srtp_ciphers_;
|
|
|
|
// Do DTLS or not
|
|
SSLMode ssl_mode_;
|
|
|
|
// Max. allowed protocol version
|
|
SSLProtocolVersion ssl_max_version_;
|
|
|
|
// A 50-ms initial timeout ensures rapid setup on fast connections, but may
|
|
// be too aggressive for low bandwidth links.
|
|
int dtls_handshake_timeout_ms_ = 50;
|
|
};
|
|
|
|
/////////////////////////////////////////////////////////////////////////////
|
|
|
|
} // namespace rtc
|
|
|
|
#endif // RTC_BASE_OPENSSL_STREAM_ADAPTER_H_
|