283 lines
12 KiB
C++
283 lines
12 KiB
C++
/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef RTC_BASE_SSL_STREAM_ADAPTER_H_
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#define RTC_BASE_SSL_STREAM_ADAPTER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "rtc_base/deprecation.h"
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#include "rtc_base/ssl_certificate.h"
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#include "rtc_base/ssl_identity.h"
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#include "rtc_base/stream.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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namespace rtc {
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// Constants for SSL profile.
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const int TLS_NULL_WITH_NULL_NULL = 0;
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const int SSL_CIPHER_SUITE_MAX_VALUE = 0xFFFF;
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// Constants for SRTP profiles.
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const int SRTP_INVALID_CRYPTO_SUITE = 0;
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#ifndef SRTP_AES128_CM_SHA1_80
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const int SRTP_AES128_CM_SHA1_80 = 0x0001;
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#endif
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#ifndef SRTP_AES128_CM_SHA1_32
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const int SRTP_AES128_CM_SHA1_32 = 0x0002;
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#endif
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#ifndef SRTP_AEAD_AES_128_GCM
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const int SRTP_AEAD_AES_128_GCM = 0x0007;
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#endif
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#ifndef SRTP_AEAD_AES_256_GCM
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const int SRTP_AEAD_AES_256_GCM = 0x0008;
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#endif
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const int SRTP_CRYPTO_SUITE_MAX_VALUE = 0xFFFF;
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// Names of SRTP profiles listed above.
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// 128-bit AES with 80-bit SHA-1 HMAC.
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extern const char CS_AES_CM_128_HMAC_SHA1_80[];
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// 128-bit AES with 32-bit SHA-1 HMAC.
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extern const char CS_AES_CM_128_HMAC_SHA1_32[];
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// 128-bit AES GCM with 16 byte AEAD auth tag.
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extern const char CS_AEAD_AES_128_GCM[];
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// 256-bit AES GCM with 16 byte AEAD auth tag.
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extern const char CS_AEAD_AES_256_GCM[];
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// Given the DTLS-SRTP protection profile ID, as defined in
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// https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
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// name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2.
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std::string SrtpCryptoSuiteToName(int crypto_suite);
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// The reverse of above conversion.
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int SrtpCryptoSuiteFromName(const std::string& crypto_suite);
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// Get key length and salt length for given crypto suite. Returns true for
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// valid suites, otherwise false.
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bool GetSrtpKeyAndSaltLengths(int crypto_suite,
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int* key_length,
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int* salt_length);
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// Returns true if the given crypto suite id uses a GCM cipher.
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bool IsGcmCryptoSuite(int crypto_suite);
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// Returns true if the given crypto suite name uses a GCM cipher.
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bool IsGcmCryptoSuiteName(const std::string& crypto_suite);
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// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
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// After SSL has been started, the stream will only open on successful
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// SSL verification of certificates, and the communication is
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// encrypted of course.
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//
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// This class was written with SSLAdapter as a starting point. It
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// offers a similar interface, with two differences: there is no
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// support for a restartable SSL connection, and this class has a
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// peer-to-peer mode.
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//
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// The SSL library requires initialization and cleanup. Static method
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// for doing this are in SSLAdapter. They should possibly be moved out
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// to a neutral class.
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enum SSLRole { SSL_CLIENT, SSL_SERVER };
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enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS };
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// Note: TLS_10, TLS_11, and DTLS_10 will all be ignored, and only
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// DTLS1_2 will be accepted, if the trial flag
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// WebRTC-LegacyTlsProtocols/Disabled/ is passed in. Support for these
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// protocol versions will be completely removed in M84 or later.
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// TODO(https://bugs.webrtc.org/10261).
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enum SSLProtocolVersion {
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SSL_PROTOCOL_NOT_GIVEN = -1,
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SSL_PROTOCOL_TLS_10 = 0,
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SSL_PROTOCOL_TLS_11,
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SSL_PROTOCOL_TLS_12,
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SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11,
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SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12,
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};
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enum class SSLPeerCertificateDigestError {
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NONE,
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UNKNOWN_ALGORITHM,
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INVALID_LENGTH,
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VERIFICATION_FAILED,
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};
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// Errors for Read -- in the high range so no conflict with OpenSSL.
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enum { SSE_MSG_TRUNC = 0xff0001 };
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// Used to send back UMA histogram value. Logged when Dtls handshake fails.
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enum class SSLHandshakeError { UNKNOWN, INCOMPATIBLE_CIPHERSUITE, MAX_VALUE };
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class SSLStreamAdapter : public StreamAdapterInterface {
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public:
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// Instantiate an SSLStreamAdapter wrapping the given stream,
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// (using the selected implementation for the platform).
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// Caller is responsible for freeing the returned object.
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static std::unique_ptr<SSLStreamAdapter> Create(
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std::unique_ptr<StreamInterface> stream);
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explicit SSLStreamAdapter(std::unique_ptr<StreamInterface> stream);
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~SSLStreamAdapter() override;
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// Specify our SSL identity: key and certificate. SSLStream takes ownership
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// of the SSLIdentity object and will free it when appropriate. Should be
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// called no more than once on a given SSLStream instance.
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virtual void SetIdentity(std::unique_ptr<SSLIdentity> identity) = 0;
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virtual SSLIdentity* GetIdentityForTesting() const = 0;
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// Call this to indicate that we are to play the server role (or client role,
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// if the default argument is replaced by SSL_CLIENT).
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// The default argument is for backward compatibility.
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// TODO(ekr@rtfm.com): rename this SetRole to reflect its new function
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virtual void SetServerRole(SSLRole role = SSL_SERVER) = 0;
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// Do DTLS or TLS.
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virtual void SetMode(SSLMode mode) = 0;
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// Set maximum supported protocol version. The highest version supported by
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// both ends will be used for the connection, i.e. if one party supports
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// DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
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// If requested version is not supported by underlying crypto library, the
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// next lower will be used.
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virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0;
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// Set the initial retransmission timeout for DTLS messages. When the timeout
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// expires, the message gets retransmitted and the timeout is exponentially
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// increased.
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// This should only be called before StartSSL().
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virtual void SetInitialRetransmissionTimeout(int timeout_ms) = 0;
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// StartSSL starts negotiation with a peer, whose certificate is verified
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// using the certificate digest. Generally, SetIdentity() and possibly
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// SetServerRole() should have been called before this.
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// SetPeerCertificateDigest() must also be called. It may be called after
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// StartSSLWithPeer() but must be called before the underlying stream opens.
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//
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// Use of the stream prior to calling StartSSL will pass data in clear text.
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// Calling StartSSL causes SSL negotiation to begin as soon as possible: right
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// away if the underlying wrapped stream is already opened, or else as soon as
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// it opens.
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//
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// StartSSL returns a negative error code on failure. Returning 0 means
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// success so far, but negotiation is probably not complete and will continue
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// asynchronously. In that case, the exposed stream will open after
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// successful negotiation and verification, or an SE_CLOSE event will be
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// raised if negotiation fails.
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virtual int StartSSL() = 0;
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// Specify the digest of the certificate that our peer is expected to use.
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// Only this certificate will be accepted during SSL verification. The
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// certificate is assumed to have been obtained through some other secure
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// channel (such as the signaling channel). This must specify the terminal
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// certificate, not just a CA. SSLStream makes a copy of the digest value.
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//
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// Returns true if successful.
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// |error| is optional and provides more information about the failure.
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virtual bool SetPeerCertificateDigest(
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const std::string& digest_alg,
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const unsigned char* digest_val,
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size_t digest_len,
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SSLPeerCertificateDigestError* error = nullptr) = 0;
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// Retrieves the peer's certificate chain including leaf certificate, if a
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// connection has been established.
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virtual std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const = 0;
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// Retrieves the IANA registration id of the cipher suite used for the
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// connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
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virtual bool GetSslCipherSuite(int* cipher_suite);
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// Retrieves the enum value for SSL version.
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// Will return -1 until the version has been negotiated.
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virtual SSLProtocolVersion GetSslVersion() const = 0;
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// Retrieves the 2-byte version from the TLS protocol.
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// Will return false until the version has been negotiated.
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virtual bool GetSslVersionBytes(int* version) const = 0;
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// Key Exporter interface from RFC 5705
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// Arguments are:
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// label -- the exporter label.
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// part of the RFC defining each exporter
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// usage (IN)
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// context/context_len -- a context to bind to for this connection;
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// optional, can be null, 0 (IN)
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// use_context -- whether to use the context value
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// (needed to distinguish no context from
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// zero-length ones).
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// result -- where to put the computed value
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// result_len -- the length of the computed value
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virtual bool ExportKeyingMaterial(const std::string& label,
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const uint8_t* context,
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size_t context_len,
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bool use_context,
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uint8_t* result,
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size_t result_len);
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// DTLS-SRTP interface
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virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites);
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virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite);
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// Returns true if a TLS connection has been established.
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// The only difference between this and "GetState() == SE_OPEN" is that if
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// the peer certificate digest hasn't been verified, the state will still be
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// SS_OPENING but IsTlsConnected should return true.
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virtual bool IsTlsConnected() = 0;
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// Capabilities testing.
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// Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now
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// that's assumed.
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static bool IsBoringSsl();
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// Returns true iff the supplied cipher is deemed to be strong.
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// TODO(torbjorng): Consider removing the KeyType argument.
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static bool IsAcceptableCipher(int cipher, KeyType key_type);
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static bool IsAcceptableCipher(const std::string& cipher, KeyType key_type);
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// TODO(guoweis): Move this away from a static class method. Currently this is
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// introduced such that any caller could depend on sslstreamadapter.h without
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// depending on specific SSL implementation.
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static std::string SslCipherSuiteToName(int cipher_suite);
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////////////////////////////////////////////////////////////////////////////
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// Testing only member functions
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////////////////////////////////////////////////////////////////////////////
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// Use our timeutils.h source of timing in BoringSSL, allowing us to test
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// using a fake clock.
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static void EnableTimeCallbackForTesting();
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// Deprecated. Do not use this API outside of testing.
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// Do not set this to false outside of testing.
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void SetClientAuthEnabledForTesting(bool enabled) {
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client_auth_enabled_ = enabled;
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}
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// Deprecated. Do not use this API outside of testing.
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// Returns true by default, else false if explicitly set to disable client
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// authentication.
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bool GetClientAuthEnabled() const { return client_auth_enabled_; }
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sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError;
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private:
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// If true (default), the client is required to provide a certificate during
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// handshake. If no certificate is given, handshake fails. This applies to
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// server mode only.
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bool client_auth_enabled_ = true;
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};
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} // namespace rtc
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#endif // RTC_BASE_SSL_STREAM_ADAPTER_H_
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