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henrike@webrtc.orgf0488722014-05-13 18:00:26 +00001/*
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
14#include <string>
15#include <vector>
16
17#include "webrtc/base/stream.h"
18#include "webrtc/base/sslidentity.h"
19
20namespace rtc {
21
Guo-wei Shieh456696a2015-09-30 21:48:54 -070022// Constants for SRTP profiles.
Guo-wei Shieh6caafbe2015-10-05 12:43:27 -070023const int SRTP_AES128_CM_SHA1_80 = 0x0001;
24const int SRTP_AES128_CM_SHA1_32 = 0x0002;
Guo-wei Shieh456696a2015-09-30 21:48:54 -070025
26// Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
27// in applications (voice) where the additional bandwidth may be significant.
28// A 80-bit HMAC is always used for SRTCP.
29// 128-bit AES with 80-bit SHA-1 HMAC.
30extern const char CS_AES_CM_128_HMAC_SHA1_80[];
31// 128-bit AES with 32-bit SHA-1 HMAC.
32extern const char CS_AES_CM_128_HMAC_SHA1_32[];
33
34// Returns the DTLS-SRTP protection profile ID, as defined in
35// https://tools.ietf.org/html/rfc5764#section-4.1.2, for the given SRTP
36// Crypto-suite, as defined in https://tools.ietf.org/html/rfc4568#section-6.2
Guo-wei Shieh6caafbe2015-10-05 12:43:27 -070037int GetSrtpCryptoSuiteFromName(const std::string& cipher_rfc_name);
Guo-wei Shieh456696a2015-09-30 21:48:54 -070038
henrike@webrtc.orgf0488722014-05-13 18:00:26 +000039// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
40// After SSL has been started, the stream will only open on successful
41// SSL verification of certificates, and the communication is
42// encrypted of course.
43//
44// This class was written with SSLAdapter as a starting point. It
45// offers a similar interface, with two differences: there is no
46// support for a restartable SSL connection, and this class has a
47// peer-to-peer mode.
48//
49// The SSL library requires initialization and cleanup. Static method
50// for doing this are in SSLAdapter. They should possibly be moved out
51// to a neutral class.
52
53
54enum SSLRole { SSL_CLIENT, SSL_SERVER };
55enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS };
Joachim Bauch831c5582015-05-20 12:48:41 +020056enum SSLProtocolVersion {
57 SSL_PROTOCOL_TLS_10,
58 SSL_PROTOCOL_TLS_11,
59 SSL_PROTOCOL_TLS_12,
60 SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11,
61 SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12,
62};
henrike@webrtc.orgf0488722014-05-13 18:00:26 +000063
64// Errors for Read -- in the high range so no conflict with OpenSSL.
65enum { SSE_MSG_TRUNC = 0xff0001 };
66
67class SSLStreamAdapter : public StreamAdapterInterface {
68 public:
69 // Instantiate an SSLStreamAdapter wrapping the given stream,
70 // (using the selected implementation for the platform).
71 // Caller is responsible for freeing the returned object.
72 static SSLStreamAdapter* Create(StreamInterface* stream);
73
74 explicit SSLStreamAdapter(StreamInterface* stream)
tkchin@webrtc.orgc569a492014-09-23 05:56:44 +000075 : StreamAdapterInterface(stream), ignore_bad_cert_(false),
76 client_auth_enabled_(true) { }
henrike@webrtc.orgf0488722014-05-13 18:00:26 +000077
78 void set_ignore_bad_cert(bool ignore) { ignore_bad_cert_ = ignore; }
79 bool ignore_bad_cert() const { return ignore_bad_cert_; }
80
tkchin@webrtc.orgc569a492014-09-23 05:56:44 +000081 void set_client_auth_enabled(bool enabled) { client_auth_enabled_ = enabled; }
82 bool client_auth_enabled() const { return client_auth_enabled_; }
83
henrike@webrtc.orgf0488722014-05-13 18:00:26 +000084 // Specify our SSL identity: key and certificate. Mostly this is
85 // only used in the peer-to-peer mode (unless we actually want to
86 // provide a client certificate to a server).
87 // SSLStream takes ownership of the SSLIdentity object and will
88 // free it when appropriate. Should be called no more than once on a
89 // given SSLStream instance.
90 virtual void SetIdentity(SSLIdentity* identity) = 0;
91
92 // Call this to indicate that we are to play the server's role in
93 // the peer-to-peer mode.
94 // The default argument is for backward compatibility
95 // TODO(ekr@rtfm.com): rename this SetRole to reflect its new function
96 virtual void SetServerRole(SSLRole role = SSL_SERVER) = 0;
97
98 // Do DTLS or TLS
99 virtual void SetMode(SSLMode mode) = 0;
100
Joachim Bauch831c5582015-05-20 12:48:41 +0200101 // Set maximum supported protocol version. The highest version supported by
102 // both ends will be used for the connection, i.e. if one party supports
103 // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
104 // If requested version is not supported by underlying crypto library, the
105 // next lower will be used.
106 virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0;
107
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000108 // The mode of operation is selected by calling either
109 // StartSSLWithServer or StartSSLWithPeer.
110 // Use of the stream prior to calling either of these functions will
111 // pass data in clear text.
112 // Calling one of these functions causes SSL negotiation to begin as
113 // soon as possible: right away if the underlying wrapped stream is
114 // already opened, or else as soon as it opens.
115 //
116 // These functions return a negative error code on failure.
117 // Returning 0 means success so far, but negotiation is probably not
118 // complete and will continue asynchronously. In that case, the
119 // exposed stream will open after successful negotiation and
120 // verification, or an SE_CLOSE event will be raised if negotiation
121 // fails.
122
123 // StartSSLWithServer starts SSL negotiation with a server in
124 // traditional mode. server_name specifies the expected server name
125 // which the server's certificate needs to specify.
126 virtual int StartSSLWithServer(const char* server_name) = 0;
127
128 // StartSSLWithPeer starts negotiation in the special peer-to-peer
129 // mode.
130 // Generally, SetIdentity() and possibly SetServerRole() should have
131 // been called before this.
132 // SetPeerCertificate() or SetPeerCertificateDigest() must also be called.
133 // It may be called after StartSSLWithPeer() but must be called before the
134 // underlying stream opens.
135 virtual int StartSSLWithPeer() = 0;
136
137 // Specify the digest of the certificate that our peer is expected to use in
138 // peer-to-peer mode. Only this certificate will be accepted during
139 // SSL verification. The certificate is assumed to have been
140 // obtained through some other secure channel (such as the XMPP
141 // channel). Unlike SetPeerCertificate(), this must specify the
142 // terminal certificate, not just a CA.
143 // SSLStream makes a copy of the digest value.
144 virtual bool SetPeerCertificateDigest(const std::string& digest_alg,
145 const unsigned char* digest_val,
146 size_t digest_len) = 0;
147
148 // Retrieves the peer's X.509 certificate, if a connection has been
149 // established. It returns the transmitted over SSL, including the entire
150 // chain. The returned certificate is owned by the caller.
151 virtual bool GetPeerCertificate(SSLCertificate** cert) const = 0;
152
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700153 // Retrieves the IANA registration id of the cipher suite used for the
154 // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
Guo-wei Shieh6caafbe2015-10-05 12:43:27 -0700155 virtual bool GetSslCipherSuite(int* cipher);
pthatcher@webrtc.org3ee4fe52015-02-11 22:34:36 +0000156
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000157 // Key Exporter interface from RFC 5705
158 // Arguments are:
159 // label -- the exporter label.
160 // part of the RFC defining each exporter
161 // usage (IN)
162 // context/context_len -- a context to bind to for this connection;
163 // optional, can be NULL, 0 (IN)
164 // use_context -- whether to use the context value
165 // (needed to distinguish no context from
166 // zero-length ones).
167 // result -- where to put the computed value
168 // result_len -- the length of the computed value
169 virtual bool ExportKeyingMaterial(const std::string& label,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200170 const uint8_t* context,
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000171 size_t context_len,
172 bool use_context,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200173 uint8_t* result,
kwiberg@webrtc.org67186fe2015-03-09 22:21:53 +0000174 size_t result_len);
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000175
176 // DTLS-SRTP interface
kwiberg@webrtc.org67186fe2015-03-09 22:21:53 +0000177 virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers);
178 virtual bool GetDtlsSrtpCipher(std::string* cipher);
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000179
180 // Capabilities testing
181 static bool HaveDtls();
182 static bool HaveDtlsSrtp();
183 static bool HaveExporter();
184
Joachim Bauch831c5582015-05-20 12:48:41 +0200185 // Returns the default Ssl cipher used between streams of this class
186 // for the given protocol version. This is used by the unit tests.
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700187 // TODO(guoweis): Move this away from a static class method.
Guo-wei Shieh6caafbe2015-10-05 12:43:27 -0700188 static int GetDefaultSslCipherForTest(SSLProtocolVersion version,
189 KeyType key_type);
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700190
191 // TODO(guoweis): Move this away from a static class method. Currently this is
192 // introduced such that any caller could depend on sslstreamadapter.h without
193 // depending on specific SSL implementation.
Guo-wei Shieh6caafbe2015-10-05 12:43:27 -0700194 static std::string GetSslCipherSuiteName(int cipher);
pthatcher@webrtc.org3ee4fe52015-02-11 22:34:36 +0000195
tkchin@webrtc.orgc569a492014-09-23 05:56:44 +0000196 private:
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000197 // If true, the server certificate need not match the configured
198 // server_name, and in fact missing certificate authority and other
199 // verification errors are ignored.
200 bool ignore_bad_cert_;
tkchin@webrtc.orgc569a492014-09-23 05:56:44 +0000201
202 // If true (default), the client is required to provide a certificate during
203 // handshake. If no certificate is given, handshake fails. This applies to
204 // server mode only.
205 bool client_auth_enabled_;
henrike@webrtc.orgf0488722014-05-13 18:00:26 +0000206};
207
208} // namespace rtc
209
210#endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_