521ed7bf022c4e30574d7970c2be5be46567f4cd |
|
19-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Convert internal representation of Srtp cryptos from string to int TBR=pthatcher@webrtc.org BUG=webrtc:5043 Review URL: https://codereview.webrtc.org/1458023002 . Cr-Commit-Position: refs/heads/master@{#10703}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
318166bed75dcbc00a7b79f715f9953aff9ffbc7 |
|
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) Reason for revert: Broke chromium fyi build. Original issue's description: > Convert internal representation of Srtp cryptos from string to int. > > Note that the coversion from int to string happens in 3 places > 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames. > 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names. > 3) stats collection also needs external names. > > External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc. > Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc. > > The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams(). > > BUG=webrtc:5043 > > Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb > Cr-Commit-Position: refs/heads/master@{#10701} TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5043 Review URL: https://codereview.webrtc.org/1455233005 Cr-Commit-Position: refs/heads/master@{#10702}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
2764e1027a08a5543e04b854a27a520801faf6eb |
|
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Convert internal representation of Srtp cryptos from string to int. Note that the coversion from int to string happens in 3 places 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames. 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names. 3) stats collection also needs external names. External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc. Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc. The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams(). BUG=webrtc:5043 Review URL: https://codereview.webrtc.org/1416673006 Cr-Commit-Position: refs/heads/master@{#10701}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types. Removes the use of uint8, etc. in favor of uint8_t. BUG=webrtc:5024 R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1362503003 . Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
6caafbe5b6b777b309a6eb90a02cf54d5106fb9b |
|
05-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Convert uint16_t to int for WebRTC cipher/crypto suite. This is a follow up CL on https://codereview.webrtc.org/1337673002 BUG= R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1377733004 . Cr-Commit-Position: refs/heads/master@{#10175}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
456696a9c1bbd586701dcca3e4b2695e419a10ba |
|
01-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Change WebRTC SslCipher to be exposed as number only This is to revert the change of https://codereview.webrtc.org/1380603005/ TBR=pthatcher@webrtc.org BUG=523033 Review URL: https://codereview.webrtc.org/1375543003 . Cr-Commit-Position: refs/heads/master@{#10126}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
27dc29b0df23eed5034f28d4d5f66ea0bb425d6c |
|
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) Reason for revert: This broke chromium.fyi bot. Original issue's description: > Change WebRTC SslCipher to be exposed as number only. > > This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting. > > For SRTP, currently it's still string internally but is reported as IANA number. > > This is used by the ongoing CL https://codereview.chromium.org/1335023002. > > BUG=523033 > > Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943 > Cr-Commit-Position: refs/heads/master@{#10124} TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=523033 Review URL: https://codereview.webrtc.org/1380603005 Cr-Commit-Position: refs/heads/master@{#10125}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
4fe3c9a77386598db9abd1f0d6983aefee9cc943 |
|
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Change WebRTC SslCipher to be exposed as number only. This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting. For SRTP, currently it's still string internally but is reported as IANA number. This is used by the ongoing CL https://codereview.chromium.org/1335023002. BUG=523033 Review URL: https://codereview.webrtc.org/1337673002 Cr-Commit-Position: refs/heads/master@{#10124}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
b6d4ec418504fd947c6f96829c73180e9487e203 |
|
17-Aug-2015 |
Torbjorn Granlund <torbjorng@google.com> |
Support generation of EC keys using P256 curve and support ECDSA certs. This CL started life here: https://webrtc-codereview.appspot.com/51189004 BUG=webrtc:4685, webrtc:4686 R=hbos@webrtc.org, juberti@webrtc.org Review URL: https://codereview.webrtc.org/1189583002 . Cr-Commit-Position: refs/heads/master@{#9718}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
831c5585c7d2b4c4442e3c1255332f1c23b6a983 |
|
20-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Allow setting maximum protocol version for SSL stream adapters. This CL adds an API to SSL stream adapters to set the maximum allowed protocol version and with that implements support for DTLS 1.2. With DTLS 1.2 the default cipher changes in the unittests as follows. BoringSSL TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA -> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 NSS TLS_RSA_WITH_AES_128_CBC_SHA -> TLS_RSA_WITH_AES_128_GCM_SHA256 BUG=chromium:428343 R=juberti@google.com Review URL: https://webrtc-codereview.appspot.com/50989004 Cr-Commit-Position: refs/heads/master@{#9232}
/external/webrtc/webrtc/base/sslstreamadapter.h
|
67186fe00cc68cbe03aa66d17fb4962458ca96d2 |
|
09-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix clang style warnings in webrtc/base Mostly this consists of marking functions with override when applicable, and moving function bodies from .h to .cc files. Not inlining virtual functions with simple bodies such as { return false; } strikes me as probably losing more in readability than we gain in binary size and compilation time, but I guess it's just like any other case where enabling a generally good warning forces us to write slightly worse code in a couple of places. BUG=163 R=kjellander@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47429004 Cr-Commit-Position: refs/heads/master@{#8656} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
3ee4fe5a940128cbfe76c8609a56c69c2aeb0175 |
|
11-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Re-land: Add API to get negotiated SSL ciphers This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer. The previously approved CL https://webrtc-codereview.appspot.com/26009004/ was reverted in https://webrtc-codereview.appspot.com/40689004/ due to compilation issues while rolling into Chromium. As the new method has landed in Chromium in https://crrev.com/bc321c76ace6e1d5a03440e554ccb207159802ec, this should be safe to land here now. BUG=3976 R=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37209004 Cr-Commit-Position: refs/heads/master@{#8343} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8343 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
2bf0e90c9d152c2b4377f710d03b1eded427c9ef |
|
07-Feb-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8275 "This CL adds an API to the SSL stream adapters and ..." I'm reverting the patch due to compilation issues. It would be great if we could make sure Chromium is ready for the patch before we land it in WebRTC. As is, we're trying to roll webrtc into Chromium and we can't (this is not the only reason though). I might reland this after the roll, depending on how that goes though. Here's an example failure: e:\b\build\slave\win_gn\build\src\jingle\glue\channel_socket_adapter_unittest.cc(77) : error C2259: 'jingle_glue::MockTransportChannel' : cannot instantiate abstract class due to following members: 'bool cricket::TransportChannel::GetSslCipher(std::string *)' : is abstract e:\b\build\slave\win_gn\build\src\third_party\webrtc\p2p\base\transportchannel.h(107) : see declaration of 'cricket::TransportChannel::GetSslCipher' ninja: build stopped: subcommand failed. > This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer. > > BUG=3976 > R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/26009004 TBR=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40689004 Cr-Commit-Position: refs/heads/master@{#8282} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8282 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
1d11c8202bd19b5dc07902107bae1d3d71575e67 |
|
06-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer. BUG=3976 R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26009004 Cr-Commit-Position: refs/heads/master@{#8275} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8275 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
c569a49a3dafdb5017961736c7715624dd059240 |
|
23-Sep-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Unit tests for SSLAdapter R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17309004 Patch from Manish Jethani <manish.jethani@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
f048872e915a3ee229044ec4bc541f6cbf9e4de1 |
|
13-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. BUG=N/A R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
e9a604accd54ab14dbf98f99ccdcf3ae1c54d27c |
|
13-May-2014 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457 > Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. > > BUG=N/A > R=andrew@webrtc.org, wu@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/12199004 TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|
2c7d1b39b9374d2bc9bda4755fd4813db66a135c |
|
12-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. BUG=N/A R=andrew@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter.h
|