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Author SHA1 Message Date
pthatcher@webrtc.org
f7bb6e723b Use new API from BoringSSL to get RFC name of cipher. 
This CL uses the new API "SSL_CIPHER_get_rfc_name" from BoringSSL to
get the RFC-compliant cipher name instead of having a custom hardcoded
list of cipher names.

BUG=none
R=juberti@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40829004

Cr-Commit-Position: refs/heads/master@{#8541}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-28 01:41:49 +00:00
pthatcher@webrtc.org
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
tommi@webrtc.org
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
pthatcher@webrtc.org
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
davidben@webrtc.org
36d5c3cb44 Leave BIO_METHOD non-const. 
This breaks building against OpenSSL upstream, which is still supported on iOS.
This reverts part of https://webrtc-codereview.appspot.com/34649004.

BUG=none
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 23:06:17 +00:00
henrike@webrtc.org
c10eceab6e Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. 
The BIO_METHODs ought to be const so they can go into rodata; BoringSSL makes
BIO_new take a const BIO_METHOD *, so there's no need for it to be non-const.
Also set SRTP_PROTECTION_PROFILE as const so we can constify those within
BoringSSL (https://boringssl-review.googlesource.com/#/c/2720/)

BUG=none
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 17:59:28 +00:00
henrike@webrtc.org
d89b69aade Fix WebRTC Win64 + BoringSSL build. 
There were many size_t to int conversions. RAND_poll and RAND_seed no longer do
anything in BoringSSL, so fix that one by removing it. Use a checked_cast for
the remaining ones.

BUG=chromium:429039
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 17:23:09 +00:00
jiayl@webrtc.org
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
andresp@webrtc.org
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
jiayl@webrtc.org
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
tkchin@webrtc.org
c569a49a3d 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
 2014-09-23 05:56:44 +00:00
jiayl@webrtc.org
11c6bde474 Specify an ECDH group for ECDHE. 
By default, OpenSSL cannot negotiate ECDHE cipher suites as a server because it
doesn't know what curve to use.

BUG=chromium:406458
TEST=Download Firefox nightly build from 2014-08-12.
  https://ftp.mozilla.org/pub/mozilla.org/firefox/nightly/2014-08-12-mozilla-central-debug/
  Point Firefox to https://apprtc.appspot.com
  Point Chrome on Android to the URL Firefox redirects to (it'll say ?r=NUMBERS at the end)
  After tapping through various permissions prompts on either side, the call goes through.

R=agl@chromium.org, henrike@webrtc.org, jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 16:14:38 +00:00
henrike@webrtc.org
4e5f65a4c6 Rebase webrtc/base with r6345 version of talk/base: 
cd webrtc/base
svn diff -r 6249:6300 http://webrtc.googlecode.com/svn/trunk/talk/base >
6300.diff
patch -p0 -i 6300.diff
ls genericslot* | xargs rm
cp ../../talk/base/sigslottester* .
manual edits of sigslottester* to get rid of talk and talk_base.

BUG=3379
TBR=jiayang

Review URL: https://webrtc-codereview.appspot.com/19649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:40:11 +00:00
henrike@webrtc.org
14abcc7322 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. 
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 16:54:44 +00:00
henrike@webrtc.org
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
perkj@webrtc.org
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
henrike@webrtc.org
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