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Bug: chromium:1072401
Change-Id: I6af81a2b2b22b5f8d7edb8fb7f66f69b866db1c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174753
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31275}
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A connection is currently deleted if it has not recevied anything for
30s. This patch adds a field trial that allows modifying this value
if no pings are outstanding.
The motivation for this is to experiment with pinging slower than
once per 30s in order to save battery.
Bug: webrtc:10282
Change-Id: I3272b9d68d44fc30379bd9a6c643db6b09766486
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175005
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31239}
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This patch adds a field trial string for the IceController
to the factory interface, the string is from the
"WebRTC-IceControllerFieldTrials" key.
This makes it possible to add new field trials
using that key as needed.
Bug: chromium:1024965
Change-Id: I50498e45da3c49b8e1d620c90c674eedc15dc16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31134}
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This version of the method is deprecated in favor of the one that takes
an url parameter.
Bug: webrtc:10198
Change-Id: I7614b9cb98217663b0e2fbf1785ae1fb1484beec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173333
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31068}
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Since there is no plan to follow-up on this, this CL removes the
field trial and the conditional logic based on it.
Bug: webrtc:11503, webrtc:10282
Change-Id: Iaf005eba6af0e23ea50456d75c5c53f37d488f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173477
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31062}
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Bug: webrtc:10198
Change-Id: I5beabba3837b92d600e2d7067954adf334adbdd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173335
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31056}
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It's not expected this will make a difference, since the packet should
be read from the queue if possible as soon as it's added to it.
But we're doing this as an added precaution in case we overlooked
something. See linked bug.
Bug: chromium:1063834
Change-Id: I7a3a6d86a97683cbcbeed5ef1aaa8090cf6bf8c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172661
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30990}
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This patch adds new enum values for different types of cellular
connections.
The new costs are currently blocked when sending to remote,
(so that arbitrary network switches does not starts occurring).
The end-game for this series to be able to distinguish between
different type of cellular connections in the ice-layer (e.g when
selecting/switching connections).
BUG: webrtc:11473
Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30970}
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Adds the missing tcptype to prflx tcp candidates as tcptype is mandatory per
RFC 6544 and if missing the candidate will contain double whitespace like this
... tcptype generation ...
and will get rejected by the internal parser
BUG=webrtc:11423
Change-Id: Id61babd85cf43d56e9e6f9bf30d4cc9e00f00f60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170442
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30959}
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We thought we had resolved this issue earlier, by reading DTLS
records in a loop. But this condition may be triggered in other cases,
such as when an internal DTLS error occurs and more DTLS records
continue to be received afterwords.
Changing this from a hard to soft error will avoid a crash (which
is happening more frequently for whatever reason) and hopefully
enable us to collect logs to debug the issue further.
Bug: chromium:1063834
Change-Id: I22c01a9e064a9db65bab38d00c62a424b5a27437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30955}
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If a STUN attribute is in the "comprehension-required" range
(0x0000-0x7FFF), and the implementation does not recognize it, this
should be treated as an error (as per RFC5389), with different behavior
depending on the type of the message received.
Bug: webrtc:9063
Change-Id: Ic31b0cdd3c26772c21d770b44fe4ee4a1b47030a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/64500
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30925}
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this is a NOP refactoring, that modify return type
of IceControllerInterface::SelectConnectionToPing to a struct
(rather than existing pair). The modification is done
so that one can safely add new return values in the struct.
Step 1) Create a typedef for return value.
- merge downstream and change it to start using new type.
Step 2) Change typedef to struct,
adding constructors from old type to new type
merge and change downstream to use "real" constructors
Step 3) remove temporary constructors
Step 4) Eat cake
Each step requires a merge downstream, with corresponding
changes there.
Bug: chromium:1024965
Change-Id: I46ad4a58426a7d33b84bc78caab55893315d25c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171874
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30921}
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This patch adds a new ForgetLearnedState() method on a Connection.
The method, puts the connection into a state similar to
when it was just created.
- write_state = STATE_WRITE_INIT
- receving = false
- throw away all pending request
- reset RttEstimate
All other state is kept unchanged.
Note: It does not trigger SignalStateChange
A subsequent patch will expose the method to the IceController.
BUG: webrtc:11463
Change-Id: I055e8cd067e1bc4fd5ad64dd10f458554dbc87e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171805
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30916}
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this is a NOP refactoring, that modify return type
of IceControllerInterface::SelectConnectionToPing to a struct
(rather than existing pair). The modification is done
so that one can safely add new return values in the struct.
Step 1) Create a typedef for return value.
- merge downstream and change it to start using new type.
Step 2) Change typedef to struct,
adding constructors from old type to new type
merge and change downstream to use "real" constructors
Step 3) remove temporary constructors
Step 4) Eat cake
Each step requires a merge downstream, with corresponding
changes there.
Bug: chromium:1024965
Change-Id: I79df9528f842ea73ca8896cedd62ad3a5cf5b767
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171807
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30914}
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this is a NOP refactoring, that modify return type
of IceControllerInterface::SelectConnectionToPing to a struct
(rather than existing pair). The modification is done
so that one can safely add new return values in the struct.
Step 1) Create a typedef for return value.
- merge downstream and change it to start using new type.
Step 2) Change typedef to struct,
adding constructors from old type to new type
merge and change downstream to use "real" constructors
Step 3) remove temporary constructors
Step 4) Eat cake
Each step requires a merge downstream, with corresponding
changes there.
Bug: chromium:1024965
Change-Id: I6ebb8658a77e0ef5c24acb382c0cb6413403c168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171691
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30902}
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this patch is a followup to https://webrtc-review.googlesource.com/c/src/+/170628
and removed the now deprecated fields {local/remote}_network_id that
is now no longer used by downstream.
BUG: webrtc:11434
Change-Id: Ia322609c0b4f07b05b8592cbca7f001a115da109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171515
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30874}
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Backwards compatible overloads are provided.
Bug: none
Change-Id: I065ad6b269fe074745f9debf68862ff70fd09628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170637
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30851}
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This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay
(previously it was "only" network_id)
The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.
OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/
BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
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Bug: None
Change-Id: I9145ba5e8ad9f80aec047227aa0a95858354fd1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169725
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30689}
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This patch fixes a problem in https://webrtc.googlesource.com/src/+/71ff07369837d6575c04ebff7002d07d6e0af25f
that when adding standard compliance validation of ufrag/pwd
accidentally broken ice renomination by introducing a new "constructor".
Bug: chromium:1044521
Change-Id: If1b18b1d728e55db9da385b37162a9cb5e61ac48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169549
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30670}
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This patch introduces 2 new field trials that make p2p_transport_channel
to send ping on network switches. The purpose of this is to reduce the
time that the peers disagre on which connection to use.
- send_ping_on_switch_ice_controlling
Send a ping from the ICE_CONTROLLING side when switching connection.
- send_ping_on_nomination_ice_controlled
Send a ping from the ICE_CONTROLLED side when a connection has been
nominated by remote side.
The extra traffic by these PINGS are considered harmless since
network switches does not happen that often.
Bug: webrtc:10273
Change-Id: Id7abe268c79ceb2404c0543849d5666466e58d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169550
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30668}
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This avoid duplication. As part of this moving the overhead calculation
to the IP address class so it's easier to find and more natural to use.
Bug: webrtc:9883
Change-Id: If4d865f445bc1a302572896932966ce30294e339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169445
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30657}
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Loosen the restrictions for ice-char by also allowing
'#' (known to break) and '_' (urlsafe base64) in addition
to the existing exceptions for '-' and '='.
Also fixes typo in log message.
BUG=chromium:1053756
Change-Id: I8f254a7c25f780276452fa3e27245b6b7ad1a3ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168943
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30596}
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Loosen the restrictions for ice-char by allowing
'-' and '='. Being spec-compliant breaks interoperability.
The spec-behaviour will be restored with a notice period.
BUG=chromium:1053756,chromium:1044521
No-Try: True
Change-Id: I880babd0869302bd713912ddfcfa48866fad32c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168820
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30560}
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find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format
Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
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The AsyncTCPSocket is an AsyncPacketSocket which means it
emulates UDP-like (packet) semantics via a TCP stream. When
sending, if the entire packet could not be written then the
packet socket should indicate it wrote the whole thing and
flush out the remaining later when the socket is available.
The WriteEvent signal was already wired up but was not getting
fired (at least with the virtual sockets) since it would not
call Send() enough on the underlying socket to get an
EWOULDBLOCK that would register the async event.
This changes AsyncTCPSocket to repeatedly call Send() on the
underlying socket until the entire packet has been written
or EWOULDBLOCK was returned.
Bug: webrtc:6655
Change-Id: I41e81e0c106c9b3e712a8a0f792d28745d93f2d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168083
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30449}
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Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport. This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).
This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.
Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
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This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
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What steps will reproduce the problem?
1. Connect a TCPPort, creating a TCPConnection
2. Disconnect the interface (e.g turn it off in android)
3. Send Ping on the TCPConnection
Crash.
The TCPConnection calls FailAndPrune when it fails to reconnect
the TCPConnection. FailAndPrune which removes the StunRequests.
When this is called from the Ping() code,
that will still access the StunRequest after the call to the Connection.
Solution: Instead of calling FailAndPrune deep down in the Ping()-stack
post a message to self to do this with a "clean" stack instead.
BUG: webrtc:11315
Change-Id: Id328b1b7c92311fa5b9adbfd2eb1dd14bf19805d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167522
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30389}
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Bug: chromium:1044521
Change-Id: I7c6fb0ba5ac918858ed65f9fe503c4de6f6acce5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167683
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30387}
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https://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-39#section-5.4
Bug: chromium:1044521
Change-Id: Ia95718437dfc270b52cdf822e861a3da7cbbab76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167281
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30375}
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See b/142641135.
Bug: None
Change-Id: I59d74b0d6c53a421d8104cc5455bab2e8dcf27d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166048
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30280}
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This patch improves handshake wrt GOOG_PING support so that
- if goog_ping_enable: sender send it's goog-ping version until it gets
STUN_BINDING_RESPONSE
- receiver only sends it's goog-ping-version if getting a
goog-ping-version in the request
This means that the overhead of STUN_ATTR_GOOG_MISC_INFO is only
- added on STUN_BINDING_REQUEST until a response is received.
- added on STUN_BINDING_RESPONSE if remote peer request it.
This is wire compatible with older versions so that
- new sender will enable GOOG_PING with new/old receiver.
- old sender will enable GOOG_PING with old receiver.
- old version will not enable GOOG_PING with new receiver
(receiver expecting sender to announce first).
BUG: webrtc:11100
Change-Id: Ib3434c593988188150f4c7506918139aaf138d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165787
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30269}
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This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
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Avoid a warning-as-error of MSVC 2019 due to a test ignoring a
[[nodiscard]] return value:
C4834: discarding return value of function with 'nodiscard' attribute
Change-Id: I6b70d85769f311814393412830f48d0d8bfef63d
Bug: webrtc:11275
Change-Id: I6b70d85769f311814393412830f48d0d8bfef63d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164467
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30226}
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BUG: webrtc:11100
Change-Id: I37a23b32b339c000cc2e88793c31732f7f1d259d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165686
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30223}
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Some messages were processed after involved objects were destructed,
a.k.a. 'use after free'.
This CL fixes that by disconnecting signals before fixture destruction,
honoring CreateChannel/DestroyChannel symmetry and following what is
done in similar test cases.
Bug: webrtc:11269
Change-Id: I122aca70a9978b752edc01e5f31583f4425f3624
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165685
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30214}
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This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
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This silences some spurious messages that were generated
by https://chromium-review.googlesource.com/c/chromium/src/+/1986070
Bug: chromium:1038754
Change-Id: I950b82c01a7e5be1f5e910b148c0b201f814f430
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164529
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30156}
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This file only defined an unused alias.
Bug: None
Change-Id: I0c731401295814e8f5dd91f41350973021efd5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155173
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30138}
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This CL propagates a "closed" signal from DTLS up to the
SCTP section of the data channel controller, where it causes
closing of all open datachannels.
Bug: chromium:1030631, webrtc:10360
Change-Id: I88bb9e1aff5c25f330edfd092ef609d4fcc3a9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162206
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30099}
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This patch introduces a new type of STUN ping,
GOOG_PING_REQUEST/RESPONSE which is similar
to a STUN_BINDING but does not transmit any values.
The Connection class automatically sends these if
no STUN attributes has changed since last call to Connection::Ping()
if the remote peer has signaled that it supports it.
BUG=webrtc:11100
Change-Id: Ib1b590f0b90ca6cb56f2eb07cd62f976e246bc8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159961
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30062}
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Bug: webrtc:11144
Change-Id: I4e75e6aec033784685de3670e880bb9f2b6ee8d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30040}
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Bug: chromium:1013564
Change-Id: Ie1bb86ed6a2a7d73fe6ee666f973d809ed05a7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#30004}
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This patch adds a new enum value in IceControllerEvent::Type,
that allows an IceController to request a recheck without
any of the predefined event occuring.
This patch is a NOP for BasicIceController.
BUG=webrtc:10647
Change-Id: Idf1d0946480437109ff272946679fef559ca7beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161047
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29976}
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This patch moves the SendBindingResponse from Port
to Connection. This is a behavioural NOP, and I don't
understand why it was in Port in the firs place!
Found when working on GOOG_PING.
BUG=webrtc:11100
Change-Id: I0466c5381f08ec4926ca3380e6914f0bc0dfcf63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161081
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29963}
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This patch estimates the connection RTT using
EventBasedExponentialMovingAverage. The half time is
set to 500 but can be modified using field trials.
This new metric is currently unused, but will
be used for exploration of of whether it can be used
instead of the existing metric.
Bug: webrtc:11140
Change-Id: I9db93e9b9eb932e3cd18935cd4ce0d90fc1cb293
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161000
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29944}
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This patch makes Connection::port() protected
and add explicit methods for the use cases instead
- network() - port()->Network()
- generation() - port()->generation()
This is done to easier mock a Connection.
BUG=webrtc:10647
Change-Id: I5b35477ed9f81d57cd871072874262d0a8af2d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160784
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29929}
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MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
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This patch is a follow up on
- https://webrtc-review.googlesource.com/c/src/+/158820
- https://webrtc-review.googlesource.com/c/src/+/158205
And makes the IceController injectable into P2PTransportChannel.
This is useful so that one can only modify the behaviour of the
the controller and still use the rest of the functionality of
P2PTransportChannel.
Bug: chromium:1024965
Change-Id: I36a1bc5cb4a60da46935ce8e4ce43e3bbbfeaf6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160188
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29882}
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