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henrike@webrtc.org47be73b2014-05-13 18:00:261/*
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_
12#define WEBRTC_BASE_SSLSTREAMADAPTER_H_
13
jbauch1286d0e2016-04-26 10:13:2214#include <memory>
henrike@webrtc.org47be73b2014-05-13 18:00:2615#include <string>
16#include <vector>
17
18#include "webrtc/base/stream.h"
19#include "webrtc/base/sslidentity.h"
20
21namespace rtc {
22
Guo-wei Shieh838c3b52015-11-19 03:41:5323// Constants for SSL profile.
24const int TLS_NULL_WITH_NULL_NULL = 0;
25
Guo-wei Shieh5acc8ec2015-10-01 04:48:5426// Constants for SRTP profiles.
Guo-wei Shieh838c3b52015-11-19 03:41:5327const int SRTP_INVALID_CRYPTO_SUITE = 0;
torbjorngd7d54cb2016-04-09 18:35:2928#ifndef SRTP_AES128_CM_SHA1_80
Guo-wei Shieh82d25852015-10-05 19:43:2729const int SRTP_AES128_CM_SHA1_80 = 0x0001;
torbjorngd7d54cb2016-04-09 18:35:2930#endif
31#ifndef SRTP_AES128_CM_SHA1_32
Guo-wei Shieh82d25852015-10-05 19:43:2732const int SRTP_AES128_CM_SHA1_32 = 0x0002;
torbjorngd7d54cb2016-04-09 18:35:2933#endif
Guo-wei Shieh5acc8ec2015-10-01 04:48:5434
35// Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
36// in applications (voice) where the additional bandwidth may be significant.
37// A 80-bit HMAC is always used for SRTCP.
38// 128-bit AES with 80-bit SHA-1 HMAC.
39extern const char CS_AES_CM_128_HMAC_SHA1_80[];
40// 128-bit AES with 32-bit SHA-1 HMAC.
41extern const char CS_AES_CM_128_HMAC_SHA1_32[];
42
Guo-wei Shieh838c3b52015-11-19 03:41:5343// Given the DTLS-SRTP protection profile ID, as defined in
44// https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
45// name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2.
46std::string SrtpCryptoSuiteToName(int crypto_suite);
47
48// The reverse of above conversion.
49int SrtpCryptoSuiteFromName(const std::string& crypto_suite);
Guo-wei Shieh5acc8ec2015-10-01 04:48:5450
henrike@webrtc.org47be73b2014-05-13 18:00:2651// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
52// After SSL has been started, the stream will only open on successful
53// SSL verification of certificates, and the communication is
54// encrypted of course.
55//
56// This class was written with SSLAdapter as a starting point. It
57// offers a similar interface, with two differences: there is no
58// support for a restartable SSL connection, and this class has a
59// peer-to-peer mode.
60//
61// The SSL library requires initialization and cleanup. Static method
62// for doing this are in SSLAdapter. They should possibly be moved out
63// to a neutral class.
64
65
66enum SSLRole { SSL_CLIENT, SSL_SERVER };
67enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS };
Joachim Bauch1bb60c82015-05-20 10:48:4168enum SSLProtocolVersion {
69 SSL_PROTOCOL_TLS_10,
70 SSL_PROTOCOL_TLS_11,
71 SSL_PROTOCOL_TLS_12,
72 SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11,
73 SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12,
74};
henrike@webrtc.org47be73b2014-05-13 18:00:2675
76// Errors for Read -- in the high range so no conflict with OpenSSL.
77enum { SSE_MSG_TRUNC = 0xff0001 };
78
79class SSLStreamAdapter : public StreamAdapterInterface {
80 public:
81 // Instantiate an SSLStreamAdapter wrapping the given stream,
82 // (using the selected implementation for the platform).
83 // Caller is responsible for freeing the returned object.
84 static SSLStreamAdapter* Create(StreamInterface* stream);
85
86 explicit SSLStreamAdapter(StreamInterface* stream)
tkchin@webrtc.org4f81cfb2014-09-23 05:56:4487 : StreamAdapterInterface(stream), ignore_bad_cert_(false),
88 client_auth_enabled_(true) { }
henrike@webrtc.org47be73b2014-05-13 18:00:2689
90 void set_ignore_bad_cert(bool ignore) { ignore_bad_cert_ = ignore; }
91 bool ignore_bad_cert() const { return ignore_bad_cert_; }
92
tkchin@webrtc.org4f81cfb2014-09-23 05:56:4493 void set_client_auth_enabled(bool enabled) { client_auth_enabled_ = enabled; }
94 bool client_auth_enabled() const { return client_auth_enabled_; }
95
henrike@webrtc.org47be73b2014-05-13 18:00:2696 // Specify our SSL identity: key and certificate. Mostly this is
97 // only used in the peer-to-peer mode (unless we actually want to
98 // provide a client certificate to a server).
99 // SSLStream takes ownership of the SSLIdentity object and will
100 // free it when appropriate. Should be called no more than once on a
101 // given SSLStream instance.
102 virtual void SetIdentity(SSLIdentity* identity) = 0;
103
104 // Call this to indicate that we are to play the server's role in
105 // the peer-to-peer mode.
106 // The default argument is for backward compatibility
107 // TODO(ekr@rtfm.com): rename this SetRole to reflect its new function
108 virtual void SetServerRole(SSLRole role = SSL_SERVER) = 0;
109
110 // Do DTLS or TLS
111 virtual void SetMode(SSLMode mode) = 0;
112
Joachim Bauch1bb60c82015-05-20 10:48:41113 // Set maximum supported protocol version. The highest version supported by
114 // both ends will be used for the connection, i.e. if one party supports
115 // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
116 // If requested version is not supported by underlying crypto library, the
117 // next lower will be used.
118 virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0;
119
henrike@webrtc.org47be73b2014-05-13 18:00:26120 // The mode of operation is selected by calling either
121 // StartSSLWithServer or StartSSLWithPeer.
122 // Use of the stream prior to calling either of these functions will
123 // pass data in clear text.
124 // Calling one of these functions causes SSL negotiation to begin as
125 // soon as possible: right away if the underlying wrapped stream is
126 // already opened, or else as soon as it opens.
127 //
128 // These functions return a negative error code on failure.
129 // Returning 0 means success so far, but negotiation is probably not
130 // complete and will continue asynchronously. In that case, the
131 // exposed stream will open after successful negotiation and
132 // verification, or an SE_CLOSE event will be raised if negotiation
133 // fails.
134
135 // StartSSLWithServer starts SSL negotiation with a server in
136 // traditional mode. server_name specifies the expected server name
137 // which the server's certificate needs to specify.
138 virtual int StartSSLWithServer(const char* server_name) = 0;
139
140 // StartSSLWithPeer starts negotiation in the special peer-to-peer
141 // mode.
142 // Generally, SetIdentity() and possibly SetServerRole() should have
143 // been called before this.
144 // SetPeerCertificate() or SetPeerCertificateDigest() must also be called.
145 // It may be called after StartSSLWithPeer() but must be called before the
146 // underlying stream opens.
147 virtual int StartSSLWithPeer() = 0;
148
149 // Specify the digest of the certificate that our peer is expected to use in
150 // peer-to-peer mode. Only this certificate will be accepted during
151 // SSL verification. The certificate is assumed to have been
152 // obtained through some other secure channel (such as the XMPP
153 // channel). Unlike SetPeerCertificate(), this must specify the
154 // terminal certificate, not just a CA.
155 // SSLStream makes a copy of the digest value.
156 virtual bool SetPeerCertificateDigest(const std::string& digest_alg,
157 const unsigned char* digest_val,
158 size_t digest_len) = 0;
159
160 // Retrieves the peer's X.509 certificate, if a connection has been
161 // established. It returns the transmitted over SSL, including the entire
kwibergb137c212016-04-06 12:15:06162 // chain.
jbauch1286d0e2016-04-26 10:13:22163 virtual std::unique_ptr<SSLCertificate> GetPeerCertificate() const = 0;
henrike@webrtc.org47be73b2014-05-13 18:00:26164
Guo-wei Shieh5acc8ec2015-10-01 04:48:54165 // Retrieves the IANA registration id of the cipher suite used for the
166 // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
Guo-wei Shieh838c3b52015-11-19 03:41:53167 virtual bool GetSslCipherSuite(int* cipher_suite);
pthatcher@webrtc.org2fb69912015-02-11 22:34:36168
torbjorngdea3f232016-03-11 08:06:47169 virtual int GetSslVersion() const = 0;
170
henrike@webrtc.org47be73b2014-05-13 18:00:26171 // Key Exporter interface from RFC 5705
172 // Arguments are:
173 // label -- the exporter label.
174 // part of the RFC defining each exporter
175 // usage (IN)
176 // context/context_len -- a context to bind to for this connection;
177 // optional, can be NULL, 0 (IN)
178 // use_context -- whether to use the context value
179 // (needed to distinguish no context from
180 // zero-length ones).
181 // result -- where to put the computed value
182 // result_len -- the length of the computed value
183 virtual bool ExportKeyingMaterial(const std::string& label,
Peter Boström07e22e62015-10-07 10:23:21184 const uint8_t* context,
henrike@webrtc.org47be73b2014-05-13 18:00:26185 size_t context_len,
186 bool use_context,
Peter Boström07e22e62015-10-07 10:23:21187 uint8_t* result,
kwiberg@webrtc.org786b6342015-03-09 22:21:53188 size_t result_len);
henrike@webrtc.org47be73b2014-05-13 18:00:26189
190 // DTLS-SRTP interface
Guo-wei Shieh838c3b52015-11-19 03:41:53191 virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites);
192 virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite);
henrike@webrtc.org47be73b2014-05-13 18:00:26193
194 // Capabilities testing
195 static bool HaveDtls();
196 static bool HaveDtlsSrtp();
197 static bool HaveExporter();
198
torbjorngdea3f232016-03-11 08:06:47199 // Returns true iff the supplied cipher is deemed to be strong.
200 // TODO(torbjorng): Consider removing the KeyType argument.
201 static bool IsAcceptableCipher(int cipher, KeyType key_type);
202 static bool IsAcceptableCipher(const std::string& cipher, KeyType key_type);
Guo-wei Shieh5acc8ec2015-10-01 04:48:54203
204 // TODO(guoweis): Move this away from a static class method. Currently this is
205 // introduced such that any caller could depend on sslstreamadapter.h without
206 // depending on specific SSL implementation.
Guo-wei Shieh838c3b52015-11-19 03:41:53207 static std::string SslCipherSuiteToName(int cipher_suite);
pthatcher@webrtc.org2fb69912015-02-11 22:34:36208
tkchin@webrtc.org4f81cfb2014-09-23 05:56:44209 private:
henrike@webrtc.org47be73b2014-05-13 18:00:26210 // If true, the server certificate need not match the configured
211 // server_name, and in fact missing certificate authority and other
212 // verification errors are ignored.
213 bool ignore_bad_cert_;
tkchin@webrtc.org4f81cfb2014-09-23 05:56:44214
215 // If true (default), the client is required to provide a certificate during
216 // handshake. If no certificate is given, handshake fails. This applies to
217 // server mode only.
218 bool client_auth_enabled_;
henrike@webrtc.org47be73b2014-05-13 18:00:26219};
220
221} // namespace rtc
222
223#endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_