Add an ICE nomination attribute. If a connection switched on the controlling side, increase the nomination value set in the attribute.
The controlled side will also be ready for re-nomination option; it will switch if a nomination comes with a higher nomination value even though it may be at a lower priority.
Plus, don't nominate or re-nominate if the nomination value at the current connection has been acknowledged.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2163403002 .
Cr-Commit-Position: refs/heads/master@{#13631}
We thought we could safely remove this, but older versions of Chrome
don't do role conflict resolution properly, so it's actually not safe
to yet.
BUG=628676
Review-Url: https://codereview.webrtc.org/2152963003
Cr-Commit-Position: refs/heads/master@{#13492}
If the currently selected connection becomes not receiving and if a backup connection
becomes strong first, we will not switch the connection until X milliseconds is passed
but the selected connection is still not receiving and the backup connection is still receiving. This will prevent the connection switching from happening too frequently.
BUG=
Review-Url: https://codereview.webrtc.org/2143653005
Cr-Commit-Position: refs/heads/master@{#13480}
we will periodically check if any network does not have any connection on it and if yes, attempt to re-gather on those networks.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2025573002 .
Cr-Commit-Position: refs/heads/master@{#13367}
Previously, we were starting a periodic timer when the local
description was set. The first connection may be created at any
time after this happens, so after creating the first connection, we
need to wait until that timer next fires before sending a ping.
Now we just start that timer (and send the first ping) immediately
after the first connection becomes pingable.
This CL also removes the "Connect" method. The only vestigal
effect of this method was to start the periodic timer, which is
now not needed since it happens automatically.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2099563004 .
Cr-Commit-Position: refs/heads/master@{#13331}
This will allow media to be sent on these pairs before a binding
response is received, shortening call setup time. However, this is only
possible if the TURN servers don't require CreatePermission when
communicating with each other.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2063823008
Cr-Commit-Position: refs/heads/master@{#13268}
Reason for revert:
Breaking webrtc builder.
Original issue's description:
> Adding IceConfig option to assume TURN/TURN candidate pairs will work.
>
> This will allow media to be sent on these pairs before a binding
> response is received, shortening call setup time. However, this is only
> possible if the TURN servers don't require CreatePermission when
> communicating with each other.
>
> R=honghaiz@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/8e6134eae4117a239de67c9a9dae8f5e3235d803
> Cr-Commit-Position: refs/heads/master@{#13263}
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
Review-Url: https://codereview.webrtc.org/2090823002
Cr-Commit-Position: refs/heads/master@{#13264}
This will allow media to be sent on these pairs before a binding
response is received, shortening call setup time. However, this is only
possible if the TURN servers don't require CreatePermission when
communicating with each other.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2063823008 .
Cr-Commit-Position: refs/heads/master@{#13263}
Writable connections are pinged at a slower rate.
The function IsPingable will filter out the writable connections.
The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.
BUG=webrtc:1161
Committed: https://crrev.com/8f7a5aad55a64f0d81b6436a22ffbdfcdcde91e0
Review-Url: https://codereview.webrtc.org/1944003002
Cr-Original-Commit-Position: refs/heads/master@{#12736}
Cr-Commit-Position: refs/heads/master@{#13241}
Add sent_ping_requests, recv_ping_responses to ConnectionInfo.
recv_ping_responses_ will be incremented when OnConnectionRequestResponse() is called.
ent_ping_requests_ will be incremented when OnConnectionRequestSent() is called.
BUG=webrtc:5695
Review-Url: https://codereview.webrtc.org/1940493002
Cr-Commit-Position: refs/heads/master@{#13001}
Reason for revert:
This will take longer time for the RTT to converge.
Need to update the RTT calculation algorithm if doing this.
Original issue's description:
> Increase the stun ping interval.
>
> Writable connections are pinged at a slower rate.
> The function IsPingable will filter out the writable connections.
> The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.
>
> BUG=webrtc:1161
>
> Committed: https://crrev.com/8f7a5aad55a64f0d81b6436a22ffbdfcdcde91e0
> Cr-Commit-Position: refs/heads/master@{#12736}
TBR=honghaiz@webrtc.org,pthatcher@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:1161
Review URL: https://codereview.webrtc.org/1982713002 .
Cr-Commit-Position: refs/heads/master@{#12762}
Writable connections are pinged at a slower rate.
The function IsPingable will filter out the writable connections.
The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.
BUG=webrtc:1161
Review-Url: https://codereview.webrtc.org/1944003002
Cr-Commit-Position: refs/heads/master@{#12736}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1920043002
Cr-Commit-Position: refs/heads/master@{#12501}
Instead of using a raw pointer output parameter. This affects
SSLStreamAdapter::GetPeerCertificate
Transport::GetRemoteSSLCertificate
TransportChannel::GetRemoteSSLCertificate
TransportController::GetRemoteSSLCertificate
WebRtcSession::GetRemoteSSLCertificate
This is a good idea in general, but will also be very convenient when
scoped_ptr is gone, since unique_ptr doesn't have an .accept() method.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1802013002
Cr-Commit-Position: refs/heads/master@{#12262}
and signaling the remote side to remove its remote candidate by setting the candidate priority to 0.
BUG=
Review URL: https://codereview.webrtc.org/1648813004
Cr-Commit-Position: refs/heads/master@{#11958}
Also change the type of "time interval" to int from uint32.
Fixed a few TODO therein. I think we should have the following convention:
1. All time delay/intervals should have type int although the time instant should have time uint32_t.
2. "interval" is preferred to "delay" if the delay will be repeated (like rescheduling).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1762863002 .
Cr-Commit-Position: refs/heads/master@{#11888}
This feature is off by default and can be turned on by setting IceConfig. When turned on, we'll choose a Turn/Turn (UDP takes higher priroity) over the other types of connections while no any connection is writable. However, when there is best connection or there is pending triggered check, those will take higher priority.
BUG=webrtc:4591
Review URL: https://codereview.webrtc.org/1577233006
Cr-Commit-Position: refs/heads/master@{#11850}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
