Relanding because the broken chromium test has been fixed:
https://chromium-review.googlesource.com/582196
This CL moves the responsibility for restricting the number of IPv6
interfaces used for ICE to BasicPortAllocator. This is the right place
to do it in the first place; it's where all the rest of the filtering
occurs. And NetworkManager shouldn't need to know about ICE limitations;
only the ICE classes should.
Part of the reason I'm doing this is that I want to add a
"max_ipv6_networks" API to RTCConfiguration, so that applications can
override the default easily (see linked bug). But that means that
PeerConnection would need to be able to call "set_max_ipv6_networks" on
the underlying object that does the filtering, and that method isn't
available on the "NetworkManager" base class. So rather than adding
another method to a place it doesn't belong, I'm moving it to the place
it does belong.
In the process, I noticed that "CompareNetworks" is inconsistent with
"SortNetworks"; the former orders interfaces alphabetically, and the
latter reverse-alphabetically. I believe this was unintentional, and
results in undesirable behavior (like "eth1" being preferred over
"eth0"), so I'm fixing it and adding a test.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Original-Commit-Position: refs/heads/master@{#19112}
Committed: ad9561404c
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Commit-Position: refs/heads/master@{#19159}
Reason for revert:
Breaks IpcNetworkManagerTest.TestMergeNetworkList, because it has built-in assumptions about network ordering that it shouldn't have. Will reland after fixing that test.
Original issue's description:
> Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
>
> This CL moves the responsibility for restricting the number of IPv6
> interfaces used for ICE to BasicPortAllocator. This is the right place
> to do it in the first place; it's where all the rest of the filtering
> occurs. And NetworkManager shouldn't need to know about ICE limitations;
> only the ICE classes should.
>
> Part of the reason I'm doing this is that I want to add a
> "max_ipv6_networks" API to RTCConfiguration, so that applications can
> override the default easily (see linked bug). But that means that
> PeerConnection would need to be able to call "set_max_ipv6_networks" on
> the underlying object that does the filtering, and that method isn't
> available on the "NetworkManager" base class. So rather than adding
> another method to a place it doesn't belong, I'm moving it to the place
> it does belong.
>
> In the process, I noticed that "CompareNetworks" is inconsistent with
> "SortNetworks"; the former orders interfaces alphabetically, and the
> latter reverse-alphabetically. I believe this was unintentional, and
> results in undesirable behavior (like "eth1" being preferred over
> "eth0"), so I'm fixing it and adding a test.
>
> BUG=webrtc:7703
>
> Review-Url: https://codereview.webrtc.org/2983213002
> Cr-Commit-Position: refs/heads/master@{#19112}
> Committed: ad9561404cTBR=zhihuang@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2984853002
Cr-Commit-Position: refs/heads/master@{#19114}
This CL moves the responsibility for restricting the number of IPv6
interfaces used for ICE to BasicPortAllocator. This is the right place
to do it in the first place; it's where all the rest of the filtering
occurs. And NetworkManager shouldn't need to know about ICE limitations;
only the ICE classes should.
Part of the reason I'm doing this is that I want to add a
"max_ipv6_networks" API to RTCConfiguration, so that applications can
override the default easily (see linked bug). But that means that
PeerConnection would need to be able to call "set_max_ipv6_networks" on
the underlying object that does the filtering, and that method isn't
available on the "NetworkManager" base class. So rather than adding
another method to a place it doesn't belong, I'm moving it to the place
it does belong.
In the process, I noticed that "CompareNetworks" is inconsistent with
"SortNetworks"; the former orders interfaces alphabetically, and the
latter reverse-alphabetically. I believe this was unintentional, and
results in undesirable behavior (like "eth1" being preferred over
"eth0"), so I'm fixing it and adding a test.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Commit-Position: refs/heads/master@{#19112}
Use rtc::SystemTimeNanos() instead of std::random_device() for PRNG seed
to avoid crashing when /dev/urandom is unavailable.
This reverts commit 3beb20720db349f651c2c04970c45b1b171c025c.
Bug: webrtc:7969
Change-Id: I5ed58a789939ee4caa99ac3abf9cab18e3e19c69
Reviewed-on: https://chromium-review.googlesource.com/572070
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19033}
This reverts commit aa41f0cfa64ece911ae2ecee83fc3190d4a42935.
Reason for revert:
Apparently, use of std::random_device() causes chromium on Linux to fail with this error:
terminating with uncaught exception of type std::__1::system_error: random_device failed to open /dev/urandom: Operation not permitted
Link to bot with failure:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Tester/builds/37563
Original change's description:
> API for periodically regathering ICE candidates
>
> Adds to the RTCConfiguration `ice_regather_interval_range` which, when
> set, specifies the randomized delay between automatic runs of ICE
> regathering. The regathering will occur on all networks and re-use the
> existing ICE ufrag/password. New connections are established once the
> candidates come back and WebRTC will automatically switch to the new
> connection that corresponds to the currently selected connection.
>
> Bug: webrtc:7969
> Change-Id: I6bbf5439a48e285f704aed9f408631cba038c82b
> Reviewed-on: https://chromium-review.googlesource.com/562505
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18978}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,steveanton@webrtc.org
No-Try: true
Bug: webrtc:7969
Change-Id: I86ef99e9f1070d3ac265398831317b68f562c614
Reviewed-on: https://chromium-review.googlesource.com/571008
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19024}
Check that ice_regather_interval_range is set only when continual
regathering is also set.
Bug: webrtc:7969
Change-Id: Ifcfeee744d817cf00914418d7e682f11528faf05
Reviewed-on: https://chromium-review.googlesource.com/569358
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19009}
Adds to the RTCConfiguration `ice_regather_interval_range` which, when
set, specifies the randomized delay between automatic runs of ICE
regathering. The regathering will occur on all networks and re-use the
existing ICE ufrag/password. New connections are established once the
candidates come back and WebRTC will automatically switch to the new
connection that corresponds to the currently selected connection.
Bug: webrtc:7969
Change-Id: I6bbf5439a48e285f704aed9f408631cba038c82b
Reviewed-on: https://chromium-review.googlesource.com/562505
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18978}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
This CL adds the flag "PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS", which will
force the creation of ports not bound to any specific network interface.
These are normally only used when network enumeration fails or is disabled,
but in some circumstances (such as the one the test case adds), they're the
only thing that works.
This will result in extra ports being gathered, which is why it's only enabled
behind a flag for now. In the future, we could probably introduce more
sophisticated "pruning" logic that would lessen the impact of the extra ports
when they're redundant, and make the flag the default.
Some other minor changes that were required to make this use case work:
* Allow a TCPPort to be used for outgoing connections even if it tries and
fails to create a server socket.
* Allow Bind to fail if being called before Connect, and the IP is an "any"
address (0.0.0.0 or ::), since this bind would have been mostly pointless
anyway.
* Prevent P2PTransprotChannel from keeping a "backup" candidate pair using
an "any address" network; we only want this for actual networks.
BUG=webrtc:7798
Review-Url: https://codereview.webrtc.org/2936553003
Cr-Commit-Position: refs/heads/master@{#18578}
The only thing the physical socket server was used for was
"Wait"/"WakeUp", but it could be replaced by a simple rtc::Event.
So, removing this dependency makes things less confusing; the fact that
VirtualSocketServer takes a PhysicalSocketServer may lead someone to
think it uses real sockets internally, when it doesn't.
BUG=None
Review-Url: https://codereview.webrtc.org/2883313003
Cr-Commit-Position: refs/heads/master@{#18172}
Instead, make the pointer to the associated socket server a
construction time const, and delete its lock.
Introduces a helper class AutoSocketServerThread for code
(mainly tests) which need a socket server associated with
the current thread.
BUG=webrtc:7501
Review-Url: https://codereview.webrtc.org/2828223002
Cr-Commit-Position: refs/heads/master@{#18047}
Add an attribute to the RTCConfiguration which can be used by specific
mobile devices so that the IPv6 ICE candidates on WiFi will not be collected.
BUG=b/35725283
Review-Url: https://codereview.webrtc.org/2731813002
Cr-Commit-Position: refs/heads/master@{#17100}
This isn't used any more so there's no point in maintaining it.
BUG=None
Review-Url: https://codereview.webrtc.org/2731673002
Cr-Commit-Position: refs/heads/master@{#17016}
This is the naming scheme we've been using for internal interfaces.
Also, this CL will introduce a PacketTransportInterface in the webrtc namespace,
which would get too easily confused with the rtc:: one:
https://codereview.webrtc.org/2675173003/
BUG=None
Review-Url: https://codereview.webrtc.org/2679103006
Cr-Commit-Position: refs/heads/master@{#16539}
This CL replaces the previously-hardcoded DTLS timeout value of 50 ms with one that is picked dynamically as 2x ICE RTT (clamped between 50 and 3000 ms to keep it reasonable).
No tests yet - I'm working on these, but sending out this CL early to get feedback. I've tested it manually and on slow networks it results in successful DTLS setup in a single attempt - instead of clogging up the slow link with multiple retransmissions as done previosly.
BUG=webrtc:7099
Review-Url: https://codereview.webrtc.org/2670113002
Cr-Commit-Position: refs/heads/master@{#16426}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.
In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using
git grep -l ' ASSERT(' | grep -v common.h | \
xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2608353003
Cr-Commit-Position: refs/heads/master@{#16041}
State and priority added to ConnectionInfo. The Connection::State enum
is replaced by IceCandidatePairState enum class.
At P2PTransportChannel::GetStats, Connection::stats is called, producing
ConnectionInfo for the connection that is then filled in with additional
values from the Connection. This is refactored so that all values are
set by Connection::stats.
RTCStatsCollector is updated to surface the ConnectionInfo stats.
BUG=webrtc:6755, chromium:633550, chromium:627816
Review-Url: https://codereview.webrtc.org/2597423003
Cr-Commit-Position: refs/heads/master@{#15870}
Reason for revert:
Breaks Chromium WebRTC FYI bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12337
The error was masked by another breaking change that was committer earlier. This is the first build showing the error.
Original issue's description:
> Make P2PTransportChannel inherit from IceTransportInternal.
>
> Make P2PTransportChannel inherit from IceTransportInternal instead of
> TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
> be separated from P2PTransportChannel.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2590063002
> Cr-Commit-Position: refs/heads/master@{#15743}
> Committed: 12749d89d9TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2594343002
Cr-Commit-Position: refs/heads/master@{#15751}
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.
BUG=none
Review-Url: https://codereview.webrtc.org/2590063002
Cr-Commit-Position: refs/heads/master@{#15743}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15453}
Reason for revert:
Deletion of transport.h broke downstream builds.
Going to reland with transport.h containing enums/etc.
Original issue's description:
> Refactoring that removes P2PTransport and DtlsTransport classes.
>
> Their base class, Transport, still exists, but it now has a more specific
> role: a helper class that applies TransportDescriptions. And is renamed
> to JsepTransport as a result.
>
> TransportController is now the entity primarily responsible for managing
> TransportChannels. It also starts storing pointers to the DTLS and ICE
> chanels separately, which will make it easier to remove
> TransportChannel/TransportChannelImpl in a subsequent CL.
>
> BUG=None
>
> Committed: https://crrev.com/bd28681d02dee8c185aeb39207e8154f0ad14a37
> Cr-Commit-Position: refs/heads/master@{#15450}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2553043004
Cr-Commit-Position: refs/heads/master@{#15452}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15450}
When the selected connection becomes not receiving and there are many connections,
If we use a round-robin fashion to ping all connections, none of the connections will
be in receiving state for sufficient long time to ensure switching connections promptly.
Triggered check will help in this situation to some extent but it may still fail to switch promptly when there are a lot of connections.
With this CL, if the selected connection is weak, once we find a writable connection on a network we start to ping it with a higher priority to keep it in receiving state.
Plus, if the selected connection is weak, we choose a shorter ping interval (900ms) for all writable connections.
BUG=b/32022719
Review-Url: https://codereview.webrtc.org/2369963004
Cr-Commit-Position: refs/heads/master@{#14991}
Introduce rtc::PacketTransportInterface. Refactor cricket::TransportChannel.
Fix signal slots parameter types in all related code.
BUG=webrtc:6531
Review-Url: https://codereview.webrtc.org/2416023002
Cr-Commit-Position: refs/heads/master@{#14778}
In the swarming test, the machines sometimes were blocked for 1-2 seconds without processing anything.
This CL makes sure that 1 second timeout is only used with fake clock.
BUG=webrtc:6500
Review-Url: https://codereview.webrtc.org/2442813002
Cr-Commit-Position: refs/heads/master@{#14756}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Prune connections based on network name.
> Previously we prune connections on the same network pointer.
> So if an IPv6 and an IPv4 network are on the same network interface, IPv4 connection won't be pruned even if an IPv6 connection with higher priority becomes writable.
>
> With this change, as long as one connection becomes writable, all connections having lower priority with the same network name will be pruned.
>
> Also simplify the implementation.
>
> BUG=webrtc:6512
>
> Committed: https://crrev.com/aae2784c1fab9d1510393dec15d76caa574e2da8
> Cr-Commit-Position: refs/heads/master@{#14593}
TBR=skvlad@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6512
Review-Url: https://codereview.webrtc.org/2412433003
Cr-Commit-Position: refs/heads/master@{#14601}
Previously we prune connections on the same network pointer.
So if an IPv6 and an IPv4 network are on the same network interface, IPv4 connection won't be pruned even if an IPv6 connection with higher priority becomes writable.
With this change, as long as one connection becomes writable, all connections having lower priority with the same network name will be pruned.
Also simplify the implementation.
BUG=webrtc:6512
Review-Url: https://codereview.webrtc.org/2395243005
Cr-Commit-Position: refs/heads/master@{#14593}
The Connection class will now blindly forward SignalReadyToSend, and
P2PTransportChannel will decide whether to forward it further (which
it was already doing).
BUG=webrtc:6448
Review-Url: https://codereview.webrtc.org/2374183005
Cr-Commit-Position: refs/heads/master@{#14462}
Reason for revert:
It caused some tests in p2ptransportchannel flaky.
Original issue's description:
> Replace RelayPort with TurnPort in p2ptransportchannel tests.
>
> Also remove the relay servers in the tests.
> Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.
>
> BUG=None
> R=deadbeef@webrtc.org
>
> Committed: https://crrev.com/c8d21712dde64c7d613d1ea56c840438505a909f
> Cr-Commit-Position: refs/heads/master@{#14441}
TBR=deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2385563002
Cr-Commit-Position: refs/heads/master@{#14443}
Also remove the relay servers in the tests.
Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.
BUG=None
R=deadbeef@webrtc.org
Review URL: https://codereview.webrtc.org/2380923002 .
Cr-Commit-Position: refs/heads/master@{#14441}
I found that, depending on when it's called, ClearGettingPorts may or
may not signal CandidatesAllocationDone, and may or may not continue
to gather more ports/candidates.
I'm fixing this inconsistency by having it always signal
CandidatesAllocationDone (if needed), and always stop gathering until
the next network change event. This makes it equivalent to
StopGettingPorts, except that it allows gathering to be restarted if
a network change occurs.
I also found that P2PTransportChannel was signaling "gathering
complete" even when continual gathering was enabled. This wasn't caught
by the unit tests due to the inconsistency of ClearGettingPorts as
described above.
Review-Url: https://codereview.webrtc.org/2124283003
Cr-Commit-Position: refs/heads/master@{#13908}
The main issue was that upon receiving a binding response with a srflx
mapped address attribute, the local candidate was not updated from local
to srflx. This means the two ICE agents view the same pair differently;
one sees it as "X<->srflx" while the other sees it as "local<->X". This
causes sub-optimal prioritization and could result in the wrong pair
being selected if using aggressive nomination.
The other issue was that TCP prflx candidates were not differentiated from
UDP prflx candidates. This lead to TCP prflx candidates prioritized above TCP
host candidates.
After fixing these issues, I was able to re-enable many disabled tests, as well
as restore the check for the candidate types of the controlled agent.
BUG=webrtc:1953,webrtc:2383
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2125823004 .
Cr-Commit-Position: refs/heads/master@{#13734}
