335204c550
Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
...
Reason for revert:
Breaks chrome.
Original issue's description:
> provide RSA2048 as per RFC
>
> BUG=webrtc:4972
>
> Committed: https://crrev.com/0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e
> Cr-Commit-Position: refs/heads/master@{#10209}
TBR=hbos@webrtc.org ,juberti@google.com ,jbauch@webrtc.org ,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1397703002
Cr-Commit-Position: refs/heads/master@{#10210}
2015-10-08 09:30:21 +00:00
0df3eb03c9
provide RSA2048 as per RFC
...
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1329493005
Cr-Commit-Position: refs/heads/master@{#10209}
2015-10-08 09:06:20 +00:00
0c4e06b4c6
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}
2015-10-07 10:23:32 +00:00
6caafbe5b6
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}
2015-10-05 19:43:33 +00:00
456696a9c1
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}
2015-10-01 04:49:02 +00:00
27dc29b0df
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}
2015-10-01 02:23:15 +00:00
4fe3c9a773
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}
2015-10-01 01:49:17 +00:00
07d09364b0
Purge nss files and dependencies.
...
This replaces https://codereview.webrtc.org/1313233005
which was reverted after triggering Chromium issues.
The only difference is that we're cleaned up dependencies
on use_openssl from the gyp file.
Since https://codereview.chromium.org/1358913003 landed,
this CL should cause no Chromium issues.
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1351503004
Cr-Commit-Position: refs/heads/master@{#10019}
2015-09-22 18:58:13 +00:00
eecbab7cd5
Roll chromium_revision a28d8d5..5482f56 (346100:347609)
...
Recent changes (https://codereview.chromium.org/1311013010 ) introduces a
dependency on WebKit (Blink) in Chromium, which forces us to start pulling
down that as well (+6GB). However Blink is about to be merged into the
Chromium repo soon anyway, so the size increase is inevitable.
Luckily, this can be removed in the next roll, if we roll past
http://crrev.com/348812
The ijar dependency was introduced in https://codereview.chromium.org/1323053003 (#347208 )
Relevant changes:
* src/third_party/boringssl/src: 12fe1b2..ac8302a
* src/third_party/libvpx: a208eca..0304cef
* src/third_party/libyuv: 3c4f573..0bc626a
* src/tools/gyp: 6ee91ad..5d01a8c
Details: a28d8d5..5482f56
/DEPS
Clang version was not updated in this roll.
R=torbjorng@webrtc.org
TBR=marpan@webrtc.org
BUG=webrtc:5005, chromium:530112
Review URL: https://codereview.webrtc.org/1305043008 .
Cr-Commit-Position: refs/heads/master@{#9956}
2015-09-16 17:19:14 +00:00
9eb1365939
Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ )
...
Reason for revert:
It looks like this broke the FYI bots. I tried updating libjingle_nacl.gyp, but the IOS build still failed because in Chrome it's configured to use NSS. See https://codereview.chromium.org/1316863012/ .
Original issue's description:
> purge nss files and dependencies
>
> BUG=webrtc:4497
>
> Committed: https://crrev.com/5647a2cf3db888195c928a1259d98f72f6ecbc15
> Cr-Commit-Position: refs/heads/master@{#9862}
TBR=tommi@webrtc.org ,kjellander@webrtc.org ,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1311843006
Cr-Commit-Position: refs/heads/master@{#9867}
2015-09-05 11:39:24 +00:00
5647a2cf3d
purge nss files and dependencies
...
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1313233005
Cr-Commit-Position: refs/heads/master@{#9862}
2015-09-04 15:12:00 +00:00
b6d4ec4185
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}
2015-08-17 12:09:10 +00:00
831c5585c7
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}
2015-05-20 10:48:24 +00:00
3ee4fe5a94
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
2015-02-11 22:35:30 +00:00
2bf0e90c9d
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
2015-02-07 11:13:18 +00:00
1d11c8202b
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
2015-02-06 19:47:39 +00:00
127ca3f8e5
Disable TestDTLSConnectWithSmallMtu on all platforms.
...
Other bots elsewhere are breaking on this test, my money is on that this
might be due to different SSL versions being used on the different bots.
This test fails on at least a couple of bots that has use_openssl=1.
R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=3910
Review URL: https://webrtc-codereview.appspot.com/25839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7403 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 07:52:03 +00:00
34f2a9ea72
Initialize SSL in unittest_main.cc.
...
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 11:36:45 +00:00
f1d751c7de
Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
...
BUG=crbug/414211
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7293
Review URL: https://webrtc-codereview.appspot.com/22739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7301 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 16:38:46 +00:00
37e1846d73
Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).
...
Breaks windows bot as it was already showing on the try jobs on the
BUG=crbug/414211
R=jiayl@webrtc.org ,juberti@webrtc.org
TBR=jiayl@webrtc.org ,juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7294 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 07:30:14 +00:00
fe1eafb71a
Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
...
BUG=crbug/414211
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7293 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 21:13:39 +00:00
fded02c164
base: disabled several base tests on Mac so that rtc_unittests can be turned back on
...
BUG=N/A
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7240 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 13:10:10 +00:00
f048872e91
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
2014-05-13 18:00:26 +00:00
e9a604accd
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
2014-05-13 08:15:48 +00:00
2c7d1b39b9
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
2014-05-12 18:03:09 +00:00