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}
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}
Collected in accordance with the spec:
https://w3c.github.io/webrtc-stats/#candidatepair-dict*
totalRoundTripTime is collected as the sum of rtt measurements, it was
previously not collected.
currentRoundTripTime is collected as the latest rtt measurement, it
was previously collected as a smoothed value, which was incorrect.
Connection is updated to collect these values which are surfaced
through ConnectionInfo.
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2719523002
Cr-Commit-Position: refs/heads/master@{#16905}
Connection::nominated() is updated to mean
(remote_nomination_ || acked_nomination_), which means both a
controlling and controlled agent can be said to be "nominated".
Previously this was (remote_nomination_ > 0) which only applies to the
controlling agent.
PortTest.TestNomination added to test nomination values and nomination
stat.
This value is surfaced through cricket::ConnectionInfo::nominated.
RTCStatsCollector uses this value in its collection of
RTCIceCandidatePairStats.
RTCStatsCollectorTest.CollectRTCIceCandidatePairStats updated to test
that ConnectionInfo::nominated is surfaced using mocks.
rtcstats_integrationtest.cc updated to expect nomination set without
using mocks.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-nominated
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2709293004
Cr-Commit-Position: refs/heads/master@{#16855}
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}
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}
If a port is not used by any channel and if it has no connection for 30
seconds, it will be removed.
Note, as long as a port is used by a transport channel, it will be kept
even if it does not have any connection. This will be beneficial to
continual gathering because new connections can be created in the future
when network changes.
BUG=
R=pthatcher@webrtc.org, zhihuang@webrtc.org
Review URL: https://codereview.webrtc.org/2171183002 .
Cr-Commit-Position: refs/heads/master@{#13567}
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.
We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.
I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2192963002 .
Cr-Commit-Position: refs/heads/master@{#13566}
If all connections on a port is destroyed, it will schedule an event
to check if it is dead after a timeout. Previously if a new connection
is created but destroyed before the event is fired, it will destroy the
port. With this change, we will not destoy it until it times out again
after the last created connection is destroyed.
BUG=
R=pthatcher@webrtc.org, zhihuang@webrtc.org
Review URL: https://codereview.webrtc.org/2184013003 .
Cr-Commit-Position: refs/heads/master@{#13563}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
Cr-Original-Commit-Position: refs/heads/master@{#13335}
Cr-Commit-Position: refs/heads/master@{#13354}
Reason for revert:
Breaks Win32/Win64 Debug bots in client.webrtc waterfall
Original issue's description:
> Add config to prune low-priority TURN ports for creating connections
> When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
>
> This effectively reduces the number of TURN candidates and connections created by TURN ports.
>
> BUG=
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
> Cr-Commit-Position: refs/heads/master@{#13335}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2111663003
Cr-Commit-Position: refs/heads/master@{#13342}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Cr-Commit-Position: refs/heads/master@{#13335}
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}
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}
If the remote side sends a candidate with the same address and port with an existing candidate,
but with a new ufrag and pwd, the local client will create a new connection from it
and destroy the old connection with the same remote address.
BUG=webrtc:5915
Review-Url: https://codereview.webrtc.org/2018693002
Cr-Commit-Position: refs/heads/master@{#13000}
This is the minumum allowed size, and will allow STUN pings to be smaller.
The unit tests on the the Gturn are also modified. A username with length of 16 bytes will be generated for Gturn only.
Review-Url: https://codereview.webrtc.org/1848083002
Cr-Commit-Position: refs/heads/master@{#12876}
if the network monitor detects it after the native code does.
Also set the network cost for ethernet, wifi, unknown, cellular network type to be 0, 10, 50, 900,
so that unknown networks will have lower precedence than known networks with low cost (like Wifi) but higher precedence than known networks with high cost.
And third, infer network type based on limited name matching in Android if there is no network monitor or network monitor did not find the type.
BUG=webrtc:5890
R=pthatcher@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1976683003 .
Cr-Commit-Position: refs/heads/master@{#12833}
Depends on this CL in order to work in Chromium:
https://codereview.chromium.org/1976673002/
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).
This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.
R=pthatcher@webrtc.org
Committed: 48e9d05f51
Review URL: https://codereview.webrtc.org/1956453003 .
Cr-Commit-Position: refs/heads/master@{#12729}
Reason for revert:
Breaks remoting_unittests. They defined their own operator== which conflicts with this one.
I'll remove the operator== in a roll CL. But until it's approved, I'm reverting this so the FYI bots will pass.
Original issue's description:
> Implement RTCConfiguration.iceCandidatePoolSize.
>
> It works by creating pooled PortAllocatorSessions which can be picked up
> by a P2PTransportChannel when needed (after a local description is set).
>
> This can optimize candidate gathering time when there is some time between
> creating a PeerConnection and setting a local description.
>
> R=pthatcher@webrtc.org
>
> Committed: 48e9d05f51TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/1972043004
Cr-Commit-Position: refs/heads/master@{#12709}
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).
This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1956453003 .
Cr-Commit-Position: refs/heads/master@{#12708}
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
NOPRESUBMIT=True
BUG=webrtc:3970
Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
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}
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.
BUG=webrtc:5155,webrtc:5670
Review URL: https://codereview.webrtc.org/1821083002
Cr-Commit-Position: refs/heads/master@{#12160}
Also include that in the stun-ping request as part of the
network-info attribute.
Change the network cost to be 16 bits.
BUG=
Review URL: https://codereview.webrtc.org/1815473002
Cr-Commit-Position: refs/heads/master@{#12110}
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}
This is a good idea in general, because it makes ownership clearer,
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/1800753003
Cr-Commit-Position: refs/heads/master@{#12002}
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.
R=tommi@webrtc.orgTBR=pthatcher@webtrc.org
BUG=4173
Review URL: https://codereview.webrtc.org/1589563003 .
Cr-Commit-Position: refs/heads/master@{#11251}
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
BUG=4173
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1577873003 .
Cr-Commit-Position: refs/heads/master@{#11247}
hasn't received anything yet. Deleting an ICE connection before it is pruned or timed out
when it hasn't received anything yet leads to ICE connections being deleted
before they have a chance to send a ping and receive a response.
BUG=
Review URL: https://codereview.webrtc.org/1544003002
Cr-Commit-Position: refs/heads/master@{#11151}
If the local port and the remote candidate's protocols do not match,
do not even try to pair them.
This avoids printing out confusing logs like
"Attempt to change a remote candidate..." in p2ptransportchannel
when two remote candidates have the same port number but different
protocols.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516613002 .
Cr-Commit-Position: refs/heads/master@{#11034}
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
