Chromium Code Reviews

Side by Side Diff: webrtc/base/opensslstreamadapter.cc

Issue 1548733002: Change DTLS default from 1.0 to 1.2 for webrtc. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: disable tests which fail under msan build. See webrtc bug 5381 Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments.
Jump to:
View unified diff |
« no previous file with comments | « talk/app/webrtc/peerconnectioninterface.h ('k') | webrtc/base/sslstreamadapterhelper.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 283 matching lines...)
294 } 294 }
295 295
296 ///////////////////////////////////////////////////////////////////////////// 296 /////////////////////////////////////////////////////////////////////////////
297 // OpenSSLStreamAdapter 297 // OpenSSLStreamAdapter
298 ///////////////////////////////////////////////////////////////////////////// 298 /////////////////////////////////////////////////////////////////////////////
299 299
300 OpenSSLStreamAdapter::OpenSSLStreamAdapter(StreamInterface* stream) 300 OpenSSLStreamAdapter::OpenSSLStreamAdapter(StreamInterface* stream)
301 : SSLStreamAdapter(stream), 301 : SSLStreamAdapter(stream),
302 state_(SSL_NONE), 302 state_(SSL_NONE),
303 role_(SSL_CLIENT), 303 role_(SSL_CLIENT),
304 ssl_read_needs_write_(false), ssl_write_needs_read_(false), 304 ssl_read_needs_write_(false),
305 ssl_(NULL), ssl_ctx_(NULL), 305 ssl_write_needs_read_(false),
306 ssl_(NULL),
307 ssl_ctx_(NULL),
306 custom_verification_succeeded_(false), 308 custom_verification_succeeded_(false),
307 ssl_mode_(SSL_MODE_TLS), 309 ssl_mode_(SSL_MODE_TLS),
308 ssl_max_version_(SSL_PROTOCOL_TLS_11) { 310 ssl_max_version_(SSL_PROTOCOL_TLS_12) {}
309 }
310 311
311 OpenSSLStreamAdapter::~OpenSSLStreamAdapter() { 312 OpenSSLStreamAdapter::~OpenSSLStreamAdapter() {
312 Cleanup(); 313 Cleanup();
313 } 314 }
314 315
315 void OpenSSLStreamAdapter::SetIdentity(SSLIdentity* identity) { 316 void OpenSSLStreamAdapter::SetIdentity(SSLIdentity* identity) {
316 ASSERT(!identity_); 317 ASSERT(!identity_);
317 identity_.reset(static_cast<OpenSSLIdentity*>(identity)); 318 identity_.reset(static_cast<OpenSSLIdentity*>(identity));
318 } 319 }
319 320
(...skipping 843 matching lines...)
1163 } 1164 }
1164 } else { 1165 } else {
1165 RTC_NOTREACHED(); 1166 RTC_NOTREACHED();
1166 return kDefaultSslEcCipher12; 1167 return kDefaultSslEcCipher12;
1167 } 1168 }
1168 } 1169 }
1169 1170
1170 } // namespace rtc 1171 } // namespace rtc
1171 1172
1172 #endif // HAVE_OPENSSL_SSL_H 1173 #endif // HAVE_OPENSSL_SSL_H
OLDNEW
« no previous file with comments | « talk/app/webrtc/peerconnectioninterface.h ('k') | webrtc/base/sslstreamadapterhelper.cc » ('j') | no next file with comments »

Powered by Google App Engine