Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}diff --git a/webrtc/rtc_base/sslstreamadapter.h b/webrtc/rtc_base/sslstreamadapter.h
new file mode 100644
index 0000000..8d85e92
--- /dev/null
+++ b/webrtc/rtc_base/sslstreamadapter.h
@@ -0,0 +1,275 @@
+/*
+ * 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 WEBRTC_RTC_BASE_SSLSTREAMADAPTER_H_
+#define WEBRTC_RTC_BASE_SSLSTREAMADAPTER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/stream.h"
+#include "webrtc/base/sslidentity.h"
+
+namespace rtc {
+
+// Constants for SSL profile.
+const int TLS_NULL_WITH_NULL_NULL = 0;
+
+// Constants for SRTP profiles.
+const int SRTP_INVALID_CRYPTO_SUITE = 0;
+#ifndef SRTP_AES128_CM_SHA1_80
+const int SRTP_AES128_CM_SHA1_80 = 0x0001;
+#endif
+#ifndef SRTP_AES128_CM_SHA1_32
+const int SRTP_AES128_CM_SHA1_32 = 0x0002;
+#endif
+#ifndef SRTP_AEAD_AES_128_GCM
+const int SRTP_AEAD_AES_128_GCM = 0x0007;
+#endif
+#ifndef SRTP_AEAD_AES_256_GCM
+const int SRTP_AEAD_AES_256_GCM = 0x0008;
+#endif
+
+// Names of SRTP profiles listed above.
+// 128-bit AES with 80-bit SHA-1 HMAC.
+extern const char CS_AES_CM_128_HMAC_SHA1_80[];
+// 128-bit AES with 32-bit SHA-1 HMAC.
+extern const char CS_AES_CM_128_HMAC_SHA1_32[];
+// 128-bit AES GCM with 16 byte AEAD auth tag.
+extern const char CS_AEAD_AES_128_GCM[];
+// 256-bit AES GCM with 16 byte AEAD auth tag.
+extern const char CS_AEAD_AES_256_GCM[];
+
+// Given the DTLS-SRTP protection profile ID, as defined in
+// https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
+// name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2.
+std::string SrtpCryptoSuiteToName(int crypto_suite);
+
+// The reverse of above conversion.
+int SrtpCryptoSuiteFromName(const std::string& crypto_suite);
+
+// Get key length and salt length for given crypto suite. Returns true for
+// valid suites, otherwise false.
+bool GetSrtpKeyAndSaltLengths(int crypto_suite, int *key_length,
+ int *salt_length);
+
+// Returns true if the given crypto suite id uses a GCM cipher.
+bool IsGcmCryptoSuite(int crypto_suite);
+
+// Returns true if the given crypto suite name uses a GCM cipher.
+bool IsGcmCryptoSuiteName(const std::string& crypto_suite);
+
+struct CryptoOptions {
+ CryptoOptions() {}
+
+ // Helper method to return an instance of the CryptoOptions with GCM crypto
+ // suites disabled. This method should be used instead of depending on current
+ // default values set by the constructor.
+ static CryptoOptions NoGcm();
+
+ // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ // if both sides enable it.
+ bool enable_gcm_crypto_suites = false;
+};
+
+// Returns supported crypto suites, given |crypto_options|.
+// CS_AES_CM_128_HMAC_SHA1_32 will be preferred by default.
+std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
+ const rtc::CryptoOptions& crypto_options);
+
+// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
+// After SSL has been started, the stream will only open on successful
+// SSL verification of certificates, and the communication is
+// encrypted of course.
+//
+// This class was written with SSLAdapter as a starting point. It
+// offers a similar interface, with two differences: there is no
+// support for a restartable SSL connection, and this class has a
+// peer-to-peer mode.
+//
+// The SSL library requires initialization and cleanup. Static method
+// for doing this are in SSLAdapter. They should possibly be moved out
+// to a neutral class.
+
+
+enum SSLRole { SSL_CLIENT, SSL_SERVER };
+enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS };
+enum SSLProtocolVersion {
+ SSL_PROTOCOL_TLS_10,
+ SSL_PROTOCOL_TLS_11,
+ SSL_PROTOCOL_TLS_12,
+ SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11,
+ SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12,
+};
+enum class SSLPeerCertificateDigestError {
+ NONE,
+ UNKNOWN_ALGORITHM,
+ INVALID_LENGTH,
+ VERIFICATION_FAILED,
+};
+
+// Errors for Read -- in the high range so no conflict with OpenSSL.
+enum { SSE_MSG_TRUNC = 0xff0001 };
+
+// Used to send back UMA histogram value. Logged when Dtls handshake fails.
+enum class SSLHandshakeError { UNKNOWN, INCOMPATIBLE_CIPHERSUITE, MAX_VALUE };
+
+class SSLStreamAdapter : public StreamAdapterInterface {
+ public:
+ // Instantiate an SSLStreamAdapter wrapping the given stream,
+ // (using the selected implementation for the platform).
+ // Caller is responsible for freeing the returned object.
+ static SSLStreamAdapter* Create(StreamInterface* stream);
+
+ explicit SSLStreamAdapter(StreamInterface* stream);
+ ~SSLStreamAdapter() override;
+
+ void set_ignore_bad_cert(bool ignore) { ignore_bad_cert_ = ignore; }
+ bool ignore_bad_cert() const { return ignore_bad_cert_; }
+
+ void set_client_auth_enabled(bool enabled) { client_auth_enabled_ = enabled; }
+ bool client_auth_enabled() const { return client_auth_enabled_; }
+
+ // Specify our SSL identity: key and certificate. SSLStream takes ownership
+ // of the SSLIdentity object and will free it when appropriate. Should be
+ // called no more than once on a given SSLStream instance.
+ virtual void SetIdentity(SSLIdentity* identity) = 0;
+
+ // Call this to indicate that we are to play the server role (or client role,
+ // if the default argument is replaced by SSL_CLIENT).
+ // The default argument is for backward compatibility.
+ // TODO(ekr@rtfm.com): rename this SetRole to reflect its new function
+ virtual void SetServerRole(SSLRole role = SSL_SERVER) = 0;
+
+ // Do DTLS or TLS.
+ virtual void SetMode(SSLMode mode) = 0;
+
+ // Set maximum supported protocol version. The highest version supported by
+ // both ends will be used for the connection, i.e. if one party supports
+ // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
+ // If requested version is not supported by underlying crypto library, the
+ // next lower will be used.
+ virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0;
+
+ // Set the initial retransmission timeout for DTLS messages. When the timeout
+ // expires, the message gets retransmitted and the timeout is exponentially
+ // increased.
+ // This should only be called before StartSSL().
+ virtual void SetInitialRetransmissionTimeout(int timeout_ms) = 0;
+
+ // StartSSL starts negotiation with a peer, whose certificate is verified
+ // using the certificate digest. Generally, SetIdentity() and possibly
+ // SetServerRole() should have been called before this.
+ // SetPeerCertificateDigest() must also be called. It may be called after
+ // StartSSLWithPeer() but must be called before the underlying stream opens.
+ //
+ // Use of the stream prior to calling StartSSL will pass data in clear text.
+ // Calling StartSSL causes SSL negotiation to begin as soon as possible: right
+ // away if the underlying wrapped stream is already opened, or else as soon as
+ // it opens.
+ //
+ // StartSSL returns a negative error code on failure. Returning 0 means
+ // success so far, but negotiation is probably not complete and will continue
+ // asynchronously. In that case, the exposed stream will open after
+ // successful negotiation and verification, or an SE_CLOSE event will be
+ // raised if negotiation fails.
+ virtual int StartSSL() = 0;
+
+ // Specify the digest of the certificate that our peer is expected to use.
+ // Only this certificate will be accepted during SSL verification. The
+ // certificate is assumed to have been obtained through some other secure
+ // channel (such as the signaling channel). This must specify the terminal
+ // certificate, not just a CA. SSLStream makes a copy of the digest value.
+ //
+ // Returns true if successful.
+ // |error| is optional and provides more information about the failure.
+ virtual bool SetPeerCertificateDigest(
+ const std::string& digest_alg,
+ const unsigned char* digest_val,
+ size_t digest_len,
+ SSLPeerCertificateDigestError* error = nullptr) = 0;
+
+ // Retrieves the peer's X.509 certificate, if a connection has been
+ // established. It returns the transmitted over SSL, including the entire
+ // chain.
+ virtual std::unique_ptr<SSLCertificate> GetPeerCertificate() const = 0;
+
+ // Retrieves the IANA registration id of the cipher suite used for the
+ // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
+ virtual bool GetSslCipherSuite(int* cipher_suite);
+
+ virtual int GetSslVersion() const = 0;
+
+ // Key Exporter interface from RFC 5705
+ // Arguments are:
+ // label -- the exporter label.
+ // part of the RFC defining each exporter
+ // usage (IN)
+ // context/context_len -- a context to bind to for this connection;
+ // optional, can be null, 0 (IN)
+ // use_context -- whether to use the context value
+ // (needed to distinguish no context from
+ // zero-length ones).
+ // result -- where to put the computed value
+ // result_len -- the length of the computed value
+ virtual bool ExportKeyingMaterial(const std::string& label,
+ const uint8_t* context,
+ size_t context_len,
+ bool use_context,
+ uint8_t* result,
+ size_t result_len);
+
+ // DTLS-SRTP interface
+ virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites);
+ virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite);
+
+ // Returns true if a TLS connection has been established.
+ // The only difference between this and "GetState() == SE_OPEN" is that if
+ // the peer certificate digest hasn't been verified, the state will still be
+ // SS_OPENING but IsTlsConnected should return true.
+ virtual bool IsTlsConnected() = 0;
+
+ // Capabilities testing.
+ // Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now
+ // that's assumed.
+ static bool IsBoringSsl();
+
+ // Returns true iff the supplied cipher is deemed to be strong.
+ // TODO(torbjorng): Consider removing the KeyType argument.
+ static bool IsAcceptableCipher(int cipher, KeyType key_type);
+ static bool IsAcceptableCipher(const std::string& cipher, KeyType key_type);
+
+ // TODO(guoweis): Move this away from a static class method. Currently this is
+ // introduced such that any caller could depend on sslstreamadapter.h without
+ // depending on specific SSL implementation.
+ static std::string SslCipherSuiteToName(int cipher_suite);
+
+ // Use our timeutils.h source of timing in BoringSSL, allowing us to test
+ // using a fake clock.
+ static void enable_time_callback_for_testing();
+
+ sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError;
+
+ private:
+ // If true, the server certificate need not match the configured
+ // server_name, and in fact missing certificate authority and other
+ // verification errors are ignored.
+ bool ignore_bad_cert_;
+
+ // If true (default), the client is required to provide a certificate during
+ // handshake. If no certificate is given, handshake fails. This applies to
+ // server mode only.
+ bool client_auth_enabled_;
+};
+
+} // namespace rtc
+
+#endif // WEBRTC_RTC_BASE_SSLSTREAMADAPTER_H_