Commit Graph

25 Commits

Author SHA1 Message Date
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