The following fields are logged:
- error, if there was an error.
- elapsed time in milliseconds
BUG=webrtc:8100
Review-Url: https://codereview.webrtc.org/2996933003
Cr-Commit-Position: refs/heads/master@{#19574}
This change extends the TurnPort tests to cover connections to
TURN servers over TLS.
As part of this, the TestTurnServer is extended to support
connections from clients over TLS.
Note that this also fixes the remaining bugs in webrtc:7562
Bug: webrtc:7584
Change-Id: If89ceae49d33417625464b5892d20eee4de7c3b5
Reviewed-on: https://chromium-review.googlesource.com/611520
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19397}
This commit makes the following changes:
1. Splits TestReconstructedServerUrl into 3 tests that individually
check the reconstructed URL for UDP IPv4, UDP IPv6, and TCP.
2. Factors out common code between protocols for release allocation and
reconstructed URL tests.
3. Provides functions for getting the expected RTT of various operations
based on the protocol used. TurnPort tests use a fake clock in part
to check tight bounds on the number of network round trips it takes
to complete operations like getting TURN candidates and trying
alternate servers. These RTTs are highly dependent on the
characteristics of the transport-layer protocol used, so these
details have been moved to dedicated functions with comments
explaining how the numbers are calculated.
Bug: webrtc:7584
Change-Id: I3b065e25446cb5ecd955f359625a35fb0ad46777
Reviewed-on: https://chromium-review.googlesource.com/611500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19395}
For ICE, we want sockets that are bound to specific network interfaces,
rather than to specific IP addresses. So, a while ago, we added a
"Network" class that gets passed into the Port constructor, in
addition to the IP address as before.
But we never finished the job of removing the IP address field, such that
a Port only guarantees something about the network interface it's
associated with, and not the specific IP address it ends up with.
This CL does that, and as a consequence, if a port ends up bound to
an IP address other than the "best" one (returned by Network::GetBestIP),
this *won't* be treated as an error.
This is relevant to Android, where even though we pass an IP address
into "Bind" as a way of identifying the network, the socket actually
gets bound using "android_setsocknetwork", which doesn't provide any
guarantees about the IP address. So, if a network interface has multiple
IPv6 addresses (for instance), we may not correctly predict the one
the OS will choose, and that's ok.
This CL also moves "SetAlternateLocalAddress" from VirtualSocket to
VirtualSocketServer, which makes for much more readable test code.
The next step, if there is one, is to pass along the Network class all
the way to SocketServer::Bind. Then the socket server could do smart
things with the network information. We could even stick a platform-
specific network handle in the Network object, such that the socket
server could use it for the binding, or for "sendmsg", for example.
See bug 7026 for more context about the sendmsg idea.
BUG=webrtc:7715
Review-Url: https://codereview.webrtc.org/2989303002
Cr-Commit-Position: refs/heads/master@{#19251}
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}
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}
The URL of the ICE server will be reconstructed by the Port and the URL
attribute is added to the cricket::Candidate struct so that we can tell
which ICE server the candidate was gathered from.
This CL only changes the native C++ code. The Java and Objc wrapper will
be created in separate CLs.
BUG=webrtc::7128
Review-Url: https://codereview.webrtc.org/2685053004
Cr-Commit-Position: refs/heads/master@{#16591}
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}
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}
The loopback range is 127.0.0.0/8, which is everything from 127.0.0.0 to
127.255.255.255.
BUG=chromium:649118
Review-Url: https://codereview.webrtc.org/2445933003
Cr-Commit-Position: refs/heads/master@{#14807}
This can be used for a certain security exploit, and doesn't have any
other practical applications we know of.
BUG=chromium:649118
Review-Url: https://codereview.webrtc.org/2440043004
Cr-Commit-Position: refs/heads/master@{#14751}
The "should I simulate EWOULDBLOCK?" determination now happens
solely in P2PTransportChannel. This also fixes a bug where the
"last packet id" was set even if no packet was sent.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2099783002 .
Cr-Commit-Position: refs/heads/master@{#13307}
The fake clock has a few advantages:
1. It lets use verify that operations take the expected number of
round trips.
2. It makes the tests faster by letting us remove the equivalent
of "Sleep(500)" all over the tests.
3. It makes the tests less flaky, because sometimes sleeping for
500ms or waiting for 1s is not enough.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2097793003 .
Cr-Commit-Position: refs/heads/master@{#13304}
When creating connections on turn port, check whether the local and remote candidates have the same IP address family, instead of checking the address family of the local socket against the remote candidate.
BUG=5871
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2083803002 .
Cr-Commit-Position: refs/heads/master@{#13269}
Reason for revert:
The Webrtc waterfall indicates that this revert is not necessary.
Original issue's description:
> Revert of Do not delete a connection in the turn port with permission error or refresh error. (patchset #6 id:260001 of https://codereview.webrtc.org/2068263003/ )
>
> Reason for revert:
> It broke webrtc builds.
>
> Original issue's description:
> > Do not delete a connection in the turn port with permission error, refresh error, or binding error.
> >
> > Even if those error happened, the connection may still be able to receive packets for a while.
> > If we delete the connections, all packets arriving will be dropped.
> >
> > BUG=webrtc:6007
> > R=deadbeef@webrtc.org, pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3d77deb29c15bfb8f794ef3413837a0ec0f0c131
> > Cr-Commit-Position: refs/heads/master@{#13262}
>
> 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.
> BUG=webrtc:6007
>
> Committed: https://crrev.com/3159ffae6b1d5cba2ad972bd3d8074ac85f2c7f9
> Cr-Commit-Position: refs/heads/master@{#13265}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6007
Review-Url: https://codereview.webrtc.org/2090073003
Cr-Commit-Position: refs/heads/master@{#13266}
Reason for revert:
It broke webrtc builds.
Original issue's description:
> Do not delete a connection in the turn port with permission error, refresh error, or binding error.
>
> Even if those error happened, the connection may still be able to receive packets for a while.
> If we delete the connections, all packets arriving will be dropped.
>
> BUG=webrtc:6007
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/3d77deb29c15bfb8f794ef3413837a0ec0f0c131
> Cr-Commit-Position: refs/heads/master@{#13262}
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.
BUG=webrtc:6007
Review-Url: https://codereview.webrtc.org/2090833002
Cr-Commit-Position: refs/heads/master@{#13265}
Even if those error happened, the connection may still be able to receive packets for a while.
If we delete the connections, all packets arriving will be dropped.
BUG=webrtc:6007
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2068263003 .
Cr-Commit-Position: refs/heads/master@{#13262}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
AllocationSequence is responsible for receiving incoming packets on
a shared UDP socket and passing them to the Port objects. TurnPort
may stop sharing UDP socket in which case it allocates a new socket.
AllocationSequence::OnReadPacket() wasn't handling that case properly
which was causing an assert in TurnPort::OnReadPacket().
BUG=webrtc:5757
R=honghaiz@webrtc.org, jiayl@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1871693004 .
Cr-Commit-Position: refs/heads/master@{#12675}
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1923163003
Cr-Commit-Position: refs/heads/master@{#12532}
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.
The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.
BUG=webrtc:5636
Review URL: https://codereview.webrtc.org/1793553002
Cr-Commit-Position: refs/heads/master@{#12019}
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}
For example, when the TURN port has an ALLOCATE_MISMATCH error.
BUG=webrtc:5432
Review URL: https://codereview.webrtc.org/1595613004
Cr-Commit-Position: refs/heads/master@{#11453}
If it still handle packets, when a ping arrives, it will pass the packet to p2ptransportchannel, eventually causing an ASSERT error there (when p2ptransportchannel tries to create a connection from the ping request from unknown address).
BUG=
Review URL: https://codereview.webrtc.org/1649493006
Cr-Commit-Position: refs/heads/master@{#11430}
This fixes an assert error in Turnport::OnSendStunPacket
BUG=webrtc:5388
Review URL: https://codereview.webrtc.org/1547373002
Cr-Commit-Position: refs/heads/master@{#11152}
This means that if a TURN server denies permission for an
unreachable address, we'll no longer ping it fruitlessly.
BUG=webrtc:4917
Review URL: https://codereview.webrtc.org/1415313004
Cr-Commit-Position: refs/heads/master@{#10789}
Do not delete the turn port entry right away when the respective
connection is deleted. The dependency on asyncinvoker has been added
in chromium libjingle-nacl.
BUG=webrtc:5120
Review URL: https://codereview.webrtc.org/1450263002
Cr-Commit-Position: refs/heads/master@{#10679}
Reason for revert:
I have to revert this unfortunately because it adds a dependency on AsyncInvoker, which is not included when building libjingle_nacl in Chromium.
AsyncInvoker needs to first be added to the list of sources in Chromium.
Original issue's description:
> Do not delete the turn port entry right away when the respective connection is deleted.
> BUG=webrtc:5120
>
> Committed: https://crrev.com/e58fe8ef0e6d959f54adee3ed77764927d3845cc
> Cr-Commit-Position: refs/heads/master@{#10641}
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5120
Review URL: https://codereview.webrtc.org/1449863002
Cr-Commit-Position: refs/heads/master@{#10649}
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}
This test only currently works because stun.l.google.com has an IPv4
address and the TURN port is created with an IPv6 address. But the test
would start failing if/when it starts providing an IPv6 address. Which
may already be happening, as indicated by a recent test failure.
Review URL: https://codereview.webrtc.org/1290233003
Cr-Commit-Position: refs/heads/master@{#9841}
