This CL removes the use of the rtc::Thread::socketserver() method
in one place.
Bug: webrtc:13145
Change-Id: I1a1b2501450788263d5280c43e4328ade46f4146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263320
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37340}
Removes all remaining usage of SetType and marks the method as
deprecated.
Bug: none
Change-Id: I98dc613978ffe7ad8a4ffd951dd974d56ed92983
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265100
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37137}
This is a partial reland of commit 794e68cf3dbc3b16ee8b12b5615d8a1622154dbd
which uses the new enum in P2PTransportChannel and BasicIceController.
Original change's description:
> Refactor IceControllerEvent
>
> This change is the first step in decoupling IceControllerEvent from
> the ICE switch reason. Further cleanup is earmarked, and will be
> landed after some internal cleanup.
>
> This change
> - adds a new enum - IceSwitchReason
> - adds a member for the new enum in IceControllerEvent
> - uses the new enum within P2PTransportChannel
> - adds methods to IceControllerInterface accepting the new enum
> - deprecates usages of the old enum in IceControllerInterface
> - adds conversion between the old and new enums for compatibility
>
> Bug: webrtc:14125, webrtc:14131
> Change-Id: I5b7201c3f631eb40db334dfeec842841a7e58174
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264140
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#37051}
Bug: webrtc:14125, webrtc:14131
Change-Id: I81b0338ae2f0560cd3df7ad9bd9af37e7c3499df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264554
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#37105}
* Add ctors for providing the type and transaction id at construction.
* Update tests to use them instead of SetType+SetTransactionID
* Make sure stun message enum types are based on uint16_t
* Mark SetTransactionID as deprecated.
* Mark SetStunMagicCookie as deprecated (unused in webrtc).
* Add SetTransactionIdForTest for the one test that uses it (might not
actually need it)
* Make StunRequest::Construct() protected.
* Add a TODO to follow up on this since construction of StunRequest
goes through an unnecessarily complex 3-step process involving
other classes and a virtual method.
Bug: none
Change-Id: Ib013e58f28e7b2b4fcb3b3e1034da31dfc93e9d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264546
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37079}
This reverts commit 794e68cf3dbc3b16ee8b12b5615d8a1622154dbd.
Reason for revert: Possibly breaks downstream project
Original change's description:
> Refactor IceControllerEvent
>
> This change is the first step in decoupling IceControllerEvent from the
> ICE switch reason. Further cleanup is earmarked, and will be landed
> after some internal cleanup.
>
> This change
> - adds a new enum - IceSwitchReason
> - adds a member for the new enum in IceControllerEvent
> - uses the new enum within P2PTransportChannel
> - adds methods to IceControllerInterface accepting the new enum
> - deprecates usages of the old enum in IceControllerInterface
> - adds conversion between the old and new enums for compatibility
>
> Bug: webrtc:14125, webrtc:14131
> Change-Id: I5b7201c3f631eb40db334dfeec842841a7e58174
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264140
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#37051}
Bug: webrtc:14125, webrtc:14131
Change-Id: Ie6c4dea0f9007f78ce2bf1ee4e6b1dbca08f3142
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264505
Auto-Submit: Sameer Vijaykar <samvi@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37059}
This change is the first step in decoupling IceControllerEvent from the
ICE switch reason. Further cleanup is earmarked, and will be landed
after some internal cleanup.
This change
- adds a new enum - IceSwitchReason
- adds a member for the new enum in IceControllerEvent
- uses the new enum within P2PTransportChannel
- adds methods to IceControllerInterface accepting the new enum
- deprecates usages of the old enum in IceControllerInterface
- adds conversion between the old and new enums for compatibility
Bug: webrtc:14125, webrtc:14131
Change-Id: I5b7201c3f631eb40db334dfeec842841a7e58174
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264140
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#37051}
This extends AlwaysValidPointer to avoid creating a unique_ptr inside it.
Bug: webrtc:13145
Change-Id: I73a4f18d0a7037b57f575b04b134e4f7eadceb79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263240
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37048}
This patch just refactors creation of P2P transport channel,
pushing down the IceTransportInit object rather than decomposing
it going down.
The IceTransportInit object will in subsequent patches be
extended with a field trial container.
Reason for splitting patch into this and subsequent is
to allow changes to internal factories.
Bug: webrtc:10335
Change-Id: Icc8b6e4142744b64d134bcb2d4a56777745db62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255602
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36215}
This cl/ changes so that the RTCTransportStats bytes/packets
sent/recevied is computed in P2PTransportChannel. Previously
they were computed by aggregating over the Connections, but that
does not work when Connections are created and destroyed.
Bug: webrtc:13769
Change-Id: Ia97dfae70b5aced897d4813ec007ba61bc032f87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36103}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This patch adds a vp preference field to RTCConfig.
DEFAULT, // No VPN preference.
ONLY_USE_VPN, // only use VPN connections.
NEVER_USE_VPN, // never use VPN connections
PREFER_VPN, // use a VPN connection if possible, i.e VPN connections sorts higher than all other connections.
AVOID_VPN, // only use VPN if there is no other connections, i.e VPN connections sorts last.
Bug: webrtc:13097
Change-Id: I3f95bdfa9134e082c7d389f803bd08facfb70262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229591
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34842}
This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
This patch fixes a bug where old candidates was
generated if doing GATHER_CONTINUALLY.
The problem was that the old port allocator session
was never stopped, and when the new sessio is created
it will attach to the network that will signal OnNetworkChanged().
The patch adds explicit stop of old sessions.
The problem was not possible to trigger using fake_network
as this "incorrectly" called SignalNetworkChanged directly
rather than after a Thread->Post() like network.cc does it.
Bug: webrtc:12210
Change-Id: Ief3f961bd97f06f4c4194ecbc3200c635ba63cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194961
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32675}
In P2PTransportChannel::OnConnectionStateChange there is
code that stop port allocation sessions if the modified
connection is stronly connected.
This means that local candidates are discarded (they are still
gathered, only not surfaced).
The implication of this is that if e.g doing a TURN allocation
slower than P2P is established, the TURN allocation will not be
added to list of local candidates => no TURN connection will be
created.
NOTE: If first connecting kRelay (only RELAY ONLY) then this
patch does matter that much...until an ICE restart happens :)
I discovered this when adding the emulated TURN server
to tests, and being surprised that the TURN allocations
never got used. These test does not (currently) use kRelay
as start.
Bug: webrtc:12210
Change-Id: I78a67201cf421b0e6fdd2ea684a00d740e063f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194141
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32647}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
This patch adds send_ping_on_selected_ice_controlling that is
the same like send_ping_on_switch_ice_controlling except that it
also sends the ping on initial selection (i.e not a switch(?)).
I.e it sends "extra" pings when selecting a connection.
This lets the peer know that this candidate pair has been selected,
which causes some peers to unblock sending on the candidate pair.
TODO deprecate send_ping_on_switch_ice_controlling in favor of this,
i.e I should have done it like this on first attempt :(
Bug: webrtc:10273
Change-Id: I0e02cd5fd6d677720c1fa863af5d7c2334d02d4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182780
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32006}
This patch adds a computed estimate on how long the ice stack
was disconnected before switching to a new connection.
The metric is currently computed as now - max(connection->last_data_recevied())
and has resonably good precision.
Bug: webrtc:11862
Change-Id: I8950d55f0eadcf164de089cdb715b4f7eed0a4c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31969}
This patch enables an IceController to use
Connection::ForgetLearnedState by returning it
in a SwitchResult, that will cause P2PTransportChannel
to call the method.
BUG: webrtc:11463
Change-Id: I098bbbd2fb2961822b165770189ac0c2225d1cb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176511
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31458}
This patch fixes a minor bug in the implementation of
surface_ice_candidates_on_ice_transport_type_changed. The existing
implementation correctly handles the surfacing, but accidentally also
set the SetNeedsIceRestartFlag, which made _next_ offer contain
a ice restart.
Modified existing testcase to verify this.
Bug: webrtc:8939
Change-Id: If566e3249296467668627e5941495f6036cbd903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176127
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31363}
This change fixes a problem where the eventual destruction of a
completed resolver sometimes doesn't happen. This is because the
destruction is posted to the network thread, and if it's destroyed
before the closure is executed, the resolver is leaked.
The fix is in three parts:
1. The resolver->Destroy call is performed on closure destruction
to make sure it will always run.
2. The closure is executed with task queue. This because the
RTC_DCHECK on thread:140 fires with the invoker_.
3. It's not possible to guarantee the context Destroy is called on due
to TaskQueue semantics. Therefore SignalThread::Destroy was changed
to accept any calling context and only requiring it's the last
public call to the object.
For unknown reasons, this leak doesn't trigger the leak checker, see
referred bugs for further investigation.
Bug: webrtc:7723, webrtc:11605, chromium:905542
Change-Id: I2681ff1d2416ccbc564974a65ac84781a9ed7aee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176125
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31359}
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}