| 381b4217cb36f434c56e399a852a0a15522a9596 |
|
04-Dec-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Ping backup connection at a slower rate
and make it configurable from the app.
Changed the decision on whether a connection is pingable:
1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection.
2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate.
Note the default behavior is the same as before.
Also cached the channel state since we are accessing it more often.
BUG=webrtc:5034
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1455033004 .
Cr-Commit-Position: refs/heads/master@{#10900}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 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/transport.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/p2p/base/transport.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/p2p/base/transport.h
|
| 2b5586774ca71b19cbebc38c96d1476ccb3dfdb4 |
|
27-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Exposing DTLS transport state from TransportChannel.
This is necessary in order to support the RTCPeerConnectionState enum in
the future, as well as a correct RTCIceConnectionState (which isn't a
combination ICE and DTLS state).
Review URL: https://codereview.webrtc.org/1414363002
Cr-Commit-Position: refs/heads/master@{#10419}
/external/webrtc/webrtc/p2p/base/transport.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/p2p/base/transport.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/p2p/base/transport.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/p2p/base/transport.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/p2p/base/transport.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/p2p/base/transport.h
|
| 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/transport.h
|
| 1f429e34180ca19a7fb98b89bacd34d42e9b01ec |
|
28-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Passing the new policy from PeerConnection RTCConfiguration to
p2ptransportchannel. This CL does not use the new policy yet.
BUG=
Review URL: https://codereview.webrtc.org/1369773003
Cr-Commit-Position: refs/heads/master@{#10092}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 04ac81f2fd8ef6680522438fac1894db5415a0ec |
|
21-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
BUG=4937
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345913004 .
Cr-Commit-Position: refs/heads/master@{#10004}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 275a2f16fd99b0f1eb43fd4ba62541af14e797c0 |
|
21-Sep-2015 |
tommi <tommi@webrtc.org> |
Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ )
Reason for revert:
Broke the Windows build:
[226/365] LINK_EMBED cc_perftests.exe
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj.rsp /c ..\..\remoting\protocol\channel_socket_adapter_unittest.cc /Foobj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj /Fdobj\remoting\remoting_unittests.cc.pdb
e:\b\build\slave\win\build\src\remoting\protocol\channel_socket_adapter_unittest.cc(36) : error C3861: 'set_readable': identifier not found
ninja: build stopped: subcommand failed.
Original issue's description:
> Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
> If a connection does not receive for 30 seconds, it will be deleted.
> BUG=
>
> Committed: https://crrev.com/ae16f8547d3b447f62f6660f13688585c6c3de15
> Cr-Commit-Position: refs/heads/master@{#10001}
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1356103002
Cr-Commit-Position: refs/heads/master@{#10002}
/external/webrtc/webrtc/p2p/base/transport.h
|
| ae16f8547d3b447f62f6660f13688585c6c3de15 |
|
21-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
If a connection does not receive for 30 seconds, it will be deleted.
BUG=
Review URL: https://codereview.webrtc.org/1351673003
Cr-Commit-Position: refs/heads/master@{#10001}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 3c089d751ede283e21e186885eaf705c3257ccd2 |
|
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 900996290c996193ac3e418f315354fd2bd0ea8a |
|
13-Jul-2015 |
honghaiz <honghaiz@webrtc.org> |
Add methods to set the ICE connection receiving_timeout values.
BUG=
Review URL: https://codereview.webrtc.org/1231913003
Cr-Commit-Position: refs/heads/master@{#9572}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 54360510ff9b7c61fc906d3ed360b06a5824bbf1 |
|
08-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add flakyness check based on the recently received packets.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1207563002 .
Cr-Commit-Position: refs/heads/master@{#9553}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 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/transport.h
|
| ae0f0ee79ecda30af4fc82eec4cdd07b7770336a |
|
05-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
Just use the less-evil version, DISALLOW_COPY_AND_ASSIGN macro.
This should help with my TODO in
https://chromium.googlesource.com/chromium/src/+/master/base/macros.h#33
Tested on Linux with the following command lines:
$ rm -rf out/
$ gn gen //out/Debug --args='is_debug=true target_cpu="x64" build_with_chromium=false'
$ ninja -C out/Debug
BUG=None
TEST=see above
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50599004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8927}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 245989b22ac6580d5a42346d7aab7b676dbc457e |
|
24-Mar-2015 |
Tommi <tommi@webrtc.org> |
Address comments from cr 43769004.
- Remove unnecessary hop to worker from OnChannelRequestSignaling_s.
- Remove now-not-needed component param.
- Update documentation.
R=juberti@webrtc.org
BUG=4444
Review URL: https://webrtc-codereview.appspot.com/42839004
Cr-Commit-Position: refs/heads/master@{#8852}
/external/webrtc/webrtc/p2p/base/transport.h
|
| 462dbcfc2a308ce898e9b70f394f14cdf1adc271 |
|
17-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Fix bug in Transport where channel_.clear() was being called without a lock.
Looks like this snuck in between misaligned braces.
Also switching to C++11 for loops, reducing lock scopes in a few places and removing locks in others.
BUG=4444
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43769004
Cr-Commit-Position: refs/heads/master@{#8765}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8765 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.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/p2p/base/transport.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/p2p/base/transport.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/p2p/base/transport.h
|
| a907e01c63b83240162ec736347df5e9faacfa2d |
|
28-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Adding constness.
Make a few member variables in the Transport class officially const so that it's clear that locking isn't needed for access. There are getters for some of these (e.g. content_name()) that don't have locking or checking, so making the variables const is at least a way to guard against regressions. Also making the clock_ member in overuse_frame_detector.h const for clarity that it doesn't require a lock for access.
No code change.
Review URL: https://webrtc-codereview.appspot.com/35949004
Cr-Commit-Position: refs/heads/master@{#8186}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8186 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 5647877b2d7b299837ed7ab8e8270d593fe5aa79 |
|
19-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| aacc23465b72151fece2e6836a7c43463d3ed41d |
|
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
(This is the 3rd try)
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 4cb3856a4d4782cc7abf228a7f01ea70812d9fb1 |
|
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.
BUG=
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 536f999e58ee7456d116afad734aa64d548f1a49 |
|
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
This is an un-revert of r7992 and r7993.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| f050791ba071eb208da4e95abc2ff21f57d0738f |
|
16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
This reverts r7992.
It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 4afb59903c2dcc893cd86a973cc16da4201e387c |
|
16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 930e004a817ed346a99ac8e56575326ca75e72aa |
|
17-Nov-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.
BUG=427555
Copied from https://webrtc-codereview.appspot.com/25959004/
R=harryjin@google.com, juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7693
Review URL: https://webrtc-codereview.appspot.com/32039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 6a782c2a46d83e09bb036d34b8c2363adc26d037 |
|
14-Nov-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
TBR=guoweis@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/25179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 312614a438c2104ccab6d0231d17604359674e15 |
|
13-Nov-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.
BUG=427555
Copied from https://webrtc-codereview.appspot.com/25959004/
R=harryjin@google.com, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/transport.h
|
| 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/transport.h
|
| 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/transport.h
|
| 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/transport.h
|