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}
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}
For every failing test, the solution was either to do a "has IPv4" check
before the test is run, or avoid depending on real network interfaces
altogether.
This specifically fixes rtc_unittests, peerconnection_unittests, and
webrtc_nonparallel_tests.
BUG=None
Review-Url: https://codereview.webrtc.org/2881973002
Cr-Commit-Position: refs/heads/master@{#18155}
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}
To simplify things, the candidate pool is only used in the first
offer/answer.
After setting a local description, the size is frozen, and changing ICE
servers won't refresh the pool.
After setting an answer, the pooled candidates are discarded.
BUG=webrtc:5180
Review-Url: https://codereview.webrtc.org/2717893003
Cr-Commit-Position: refs/heads/master@{#17178}
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}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
This change moves the definition of the UMA MetricsObserverInterface from api/peerconnectioninterface.h into api/umametrics.h. This allows us to remove the unwanted dependency on peerconnectioninterface.h from files in webrtc/p2p.
This is a simple refactoring with no functional changes.
BUG=None
Review-Url: https://codereview.webrtc.org/2627093005
Cr-Commit-Position: refs/heads/master@{#16020}
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
This isn't as good as being able to enumerate all networks, but it's better
than doing nothing; it still will provide STUN/TURN candidates for the default
route if one exists.
BUG=webrtc:6932
Review-Url: https://codereview.webrtc.org/2599673003
Cr-Commit-Position: refs/heads/master@{#15766}
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.
Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 1) The policy is always explicitly initialized to secure.
> 2) API users have to explicitly integrate with the feature to
> use it, and will otherwise get no change in behavior.
> 3) The feature is not immediately exposed in non-native
> contexts. For example, disabling of certificate validation
> is not implemented via URI parsing since this would
> immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9eTBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS.
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
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}
Previously when a Turn port is pruned, if its candidate has been sent to the remote side, the remote side will keep the candidate and use that to create connections.
We now signal the remote side to remove the candidates so that at least no new connection will be created using the removed candidates.
Also updated the virtual socket server to better support our test cases.
1. Allow the virtual socket server to set transit delay for packets sent from a given IP address.
2. Ensure the ordered packet delivery for each socket (Previously the delivery order is enforced on the whole test case, so if a udp packet gets delayed based on its IP address, all TCP packets sent after the UDP packet will be delayed at least until the UDP packet is received).
BUG=webrtc:6380
R=deadbeef@webrtc.org, pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2261523004 .
Cr-Commit-Position: refs/heads/master@{#14297}
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 added logs will be helpful for debugging.
If a session has stopped, terminate DoAllocate early.
Session::init always returns true, so there is no need to check the return value.
R=deadbeef@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2267163002 .
Cr-Commit-Position: refs/heads/master@{#13871}
In the existing code, if the server address and the local IP family does not
match, we still create a TurnPort and destroy it later.
Instead, we could avoid creating it at the beginning.
This does not affect the client behavior except for the port creation.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org, zhihuang@webrtc.org
Review URL: https://codereview.webrtc.org/2206713004 .
Cr-Commit-Position: refs/heads/master@{#13720}
It was possible that "A < B" and "B < A" both evaluated to true.
This manifested as an assert on Windows, and a memory leak on Linux.
Note that the concept of "less than" is meaningless for this object.
The operator is only needed so the object can be used as a key in an
std::map.
BUG=webrtc:6068
R=honghaiz@webrtc.org, kjellander@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2187913002 .
Cr-Commit-Position: refs/heads/master@{#13598}
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}
Logging when a candidate is gathered or the gathering state or a
Port changes. This will make it easier to identify problems related
to candidate gathering.
Review-Url: https://codereview.webrtc.org/2122373004
Cr-Commit-Position: refs/heads/master@{#13490}
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}
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}
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}
