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Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)

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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}
This commit is contained in:
Henrik Kjellander
2017-06-29 08:03:04 +02:00
parent ec78f1cebc
commit c03627683f
383 changed files with 1607 additions and 1467 deletions
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275
webrtc/base/sslstreamadapter.h
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@ -1,275 +0,0 @@
/*
* 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_BASE_SSLSTREAMADAPTER_H_
#define WEBRTC_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_BASE_SSLSTREAMADAPTER_H_