History log of /external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
3542013f587f0858fb24fa8e554ec3c01a323da8 14-Jan-2016 sprang <sprang@webrtc.org> Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )

Reason for revert:
We're getting boringssl version conflicts. Reverting for now.

Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org, henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}

TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1586183002

Cr-Commit-Position: refs/heads/master@{#11253}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
31c8d2eac5aec977f584ab0ae5a1d457d674f101 14-Jan-2016 Torbjorn Granlund <torbjorng@google.com> Update with new default boringssl no-aes cipher suites. Re-enable tests.

This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).

BUG=webrtc:5381
R=davidben@webrtc.org, henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1550773002 .

Cr-Commit-Position: refs/heads/master@{#11250}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
a7446d2a50167602b04f58c917f5075ad5e494dc 12-Jan-2016 Guo-wei Shieh <guoweis@webrtc.org> Change DTLS default from 1.0 to 1.2 for webrtc.

This changes for standalone webrtc applications.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1548733002 .

Cr-Commit-Position: refs/heads/master@{#11211}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
2f042f26a3d0c062c43dc553058a286bd4dd8f19 20-Dec-2015 kjellander <kjellander@webrtc.org> Roll chromium_revision 1b6c421..db567a8 (365999:366304)

I had to disable some Dtls12Both tests failing under MSan (see bug).
Notice those errors started happening in the range of
https://boringssl.googlesource.com/boringssl.git/+log/afd565f..9f897b2
while this CL brings in an even newer BoringSSL (that still has the same problem).

Change log: https://chromium.googlesource.com/chromium/src/+log/1b6c421..db567a8
Full diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/afd565f..afe57cb
* src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/1019e45..1ccbf8f
* src/third_party/nss: https://chromium.googlesource.com/chromium/deps/nss.git/+log/a676aa0..aee1b12
DEPS diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8/DEPS

No update to Clang.

NOTRY=True
BUG=webrtc:5381
TBR=torbjorng@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1533253002

Cr-Commit-Position: refs/heads/master@{#11095}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
0eb15ed7b806125774bd13fb214aeb403e2c6857 17-Dec-2015 kwiberg <kwiberg@webrtc.org> Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

We can now use std::move instead!

This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.

Review URL: https://codereview.webrtc.org/1460043002

Cr-Commit-Position: refs/heads/master@{#11064}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 15-Oct-2015 stefan <stefan@webrtc.org> Wire up packet_id / send time callbacks to webrtc via libjingle.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
c4d3a5d44c25fb42c26393b6ddc0feadd52e5e2f 30-Sep-2015 Taylor Brandstetter <deadbeef@webrtc.org> Thinning out the Transport class.

Connecting TransportChannelImpls directly to the TransportController,
and removing redundant signal forwarding/state aggregating code from
Transport. This brings us closer to just getting rid of Transport
entirely.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1380563002 .

Cr-Commit-Position: refs/heads/master@{#10120}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
cbecd358e032021eac11fb13e04ec7f070d4f407 23-Sep-2015 deadbeef <deadbeef@webrtc.org> Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )

Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/

Which fixes the FYI bots for the original CL, and breaks them for this revert.

Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}

TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1361773005

Cr-Commit-Position: refs/heads/master@{#10036}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
a81a42f584baa0d93a4b93da9632415e8922450c 23-Sep-2015 torbjorng <torbjorng@webrtc.org> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )

Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.

Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1358413003

Cr-Commit-Position: refs/heads/master@{#10024}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
47ee2f3b9f33e8938948c482c921d4e13a3acd83 23-Sep-2015 deadbeef <deadbeef@webrtc.org> TransportController refactoring.

Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

Review URL: https://codereview.webrtc.org/1350523003

Cr-Commit-Position: refs/heads/master@{#10022}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
8902433a43bbc9cc0de4966774d3dbbe37ef96fb 18-Sep-2015 Guo-wei Shieh <guoweis@webrtc.org> Revert "TransportController refactoring."

This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.

Cr-Commit-Position: refs/heads/master@{#9994}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
9af63f473e1d0d6c47a741a046c41642dfc1c178 18-Sep-2015 deadbeef <deadbeef@webrtc.org> TransportController refactoring.

Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".

Review URL: https://codereview.webrtc.org/1246913005

Cr-Commit-Position: refs/heads/master@{#9993}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
7cbd188c5ed7df80bb737bd4ada94422730e2d89 18-Sep-2015 Peter Thatcher <pthatcher@chromium.org> Remove GICE (again).

R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1353713002 .

Cr-Commit-Position: refs/heads/master@{#9979}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
d12140a68efdcffa1c2c18f25149905e9dae1a9c 10-Sep-2015 guoweis <guoweis@webrtc.org> Revert change which removes GICE.

There are still dependencies on this functionality.

TBR=pthatcher@webrtc.org

BUG=526399

Review URL: https://codereview.webrtc.org/1336553003

Cr-Commit-Position: refs/heads/master@{#9920}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
f3ecdb981c172cdfafbe92c939eb25ddcc1ae96e 08-Sep-2015 Henrik Boström <hbos@webrtc.org> Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer.

BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org

Review URL: https://codereview.webrtc.org/1304043008 .

Cr-Commit-Position: refs/heads/master@{#9885}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
d82819892a382899a82ced756a9922a84ca9ca98 27-Aug-2015 Henrik Boström <hbos@webrtc.org> Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer.

Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.

This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.

BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org

Review URL: https://codereview.webrtc.org/1312643004 .

Cr-Commit-Position: refs/heads/master@{#9794}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
2159b89fa2cb55beeef38f72bd45e217f3d33d4e 22-Aug-2015 Peter Thatcher <pthatcher@chromium.org> Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.

This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.

Original CL: https://codereview.webrtc.org/1263663002/

R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1303393002 .

Cr-Commit-Position: refs/heads/master@{#9761}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
5bdafd44c86ee46bd7e040f19828324583418b33 21-Aug-2015 minyuel <minyue@webrtc.org> Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""

This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.

Original code review see
https://codereview.webrtc.org/1291363005

The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137

TBR=pthatcher@webrtc.org,

BUG=

Review URL: https://codereview.webrtc.org/1308753003 .

Cr-Commit-Position: refs/heads/master@{#9756}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
081f34b564e1a26ffbbe9515eba1fef7c736fdde 20-Aug-2015 Peter Thatcher <pthatcher@chromium.org> Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."

This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.

R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1291363005 .

Cr-Commit-Position: refs/heads/master@{#9738}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
fa301809b698017455847f45cc7e0dfa1bdfed35 11-Aug-2015 pthatcher <pthatcher@webrtc.org> Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.

This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.

TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1274273005

Cr-Commit-Position: refs/heads/master@{#9698}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
3449faa553ec94c52ef2d0949867befb60992c88 10-Aug-2015 Peter Thatcher <pthatcher@chromium.org> Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).

R=deadbeef@webrtc.org, juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1263663002 .

Cr-Commit-Position: refs/heads/master@{#9692}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
04e5b498278c633bc3c49da43d08c15b1e75ebc0 29-May-2015 Joachim Bauch <jbauch@webrtc.org> Make maximum SSL version configurable through PeerConnectionFactory::Options

This can be used to activate DTLS 1.2 through a command-line flag from Chromium
later.

BUG=chromium:428343
R=jiayl@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/54509004

Cr-Commit-Position: refs/heads/master@{#9328}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
1b794d56b72853fe2eecbea2c719886c44578ed8 12-May-2015 Joachim Bauch <jbauch@webrtc.org> Switch to use SHA-256 for certificates / fingerprints.

This CL changes identity generation to use SHA-256 for the self-signed
certificates and the fingerprints sent in the SDP.

BUG=4602
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/47149004

Cr-Commit-Position: refs/heads/master@{#9173}
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
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/p2p/base/dtlstransportchannel_unittest.cc
269fb4bc90b79bebbb8311da0110ccd6803fd0a8 28-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
28100cb38896fe298b6df11ffd31838d9faf5b8a 18-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."

BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc
d1ba6d9cbfc44618d2c553ff7851948c730ae37b 15-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/dtlstransportchannel_unittest.cc