This is to aid with catching issues whereby a connection object might
have a bad reference back to a port object, e.g. inside of an async
callback.
Bug: webrtc:13892
Change-Id: I56503fedc2865919713b10f236ce023554c68ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257164
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36394}
convert rtc_base/network and collateral.
This also remove last usage of system_wrappers/field_trials
in p2p/...Yay!
Bug: webrtc:10335
Change-Id: Ie8507b1f52bf7f3067e9b4bf8c81a825e4644fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256640
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@{#36357}
The AsyncListenSocket::SetOption method then gets unused, and can be
deleted.
Bug: webrtc:13065
Change-Id: Idcf70a75b96036290fdceff6e0f96a8d5617f87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35302}
This is a reland of b141c162ee2ef88a7498ba8cb8bc852287f93ad2
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
Bug: webrtc:13065
Change-Id: I88bebdd80ebe6bcf6ac635023924d79fbfb76813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35260}
This reverts commit b141c162ee2ef88a7498ba8cb8bc852287f93ad2.
Reason for revert: Breaking WebRTC rolls. See https://ci.chromium.org/ui/b/8832847811929676465 for an example failed build.
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13065
Change-Id: Id5d5b35cb21704ca4e3006caf1636906df062609
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235824
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35249}
A preparation for splitting server sockets out into a separate
interface, see https://webrtc-review.googlesource.com/c/src/+/232607.
Transition plan:
1. Land this cl.
2. Update downstream code to use the new name.
3. Attempt landing
https://webrtc-review.googlesource.com/c/src/+/232607. May need
additional steps to not break downstream implementations of
PacketSocketFactory::CreateServerTcpSocket.
Bug: webrtc:13065
Change-Id: Ife448c705222f4c9f66a096e3dc7eb07e0f9c3af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35155}
This feature is used only by chromium, and only for UDP sockets.
Bug: webrtc:13065
Change-Id: I207ea643aa57cf23bdd36266895f65f1ee251aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35106}
Add a timestamp for last data sent in Connection.
Move calling of rtc::TimeMillis() to Connection and remove it from RateTracker::AddSamples.
This timestamp will be used to further improve fail over logic.
BUG=None
Change-Id: I4cbc7693a0e081277590b9cb13264dc2a998202e
No-Try: True
Change-Id: I4cbc7693a0e081277590b9cb13264dc2a998202e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197421
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32831}
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}
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}
Bug: None
Change-Id: I0e7e76ec2d61a1e2719975701a32c1cfc04f97d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Drake <alexdrake@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29103}
I gave up on removing proxy_info, user_agent and tcp_options. I don't
think it's feasible to remove them without removing all the proxy code.
The assumption that you can set the proxy and user agent long after
you have created the factory is entrenched in unit tests and the code
itself. So is the ability to set tcp opts depending on protocol or
endpoint properties.
It may be easier to untangle proxy stuff from the factory later,
when it becomes a more first-class citizen and isn't passed via
the allocator.
Requires https://chromium-review.googlesource.com/c/chromium/src/+/1778870
to land first.
Bug: webrtc:7447
Change-Id: Ib496e2bb689ea415e9f8ec1dfedff13a83fa4a8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150799
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29091}
TCPPort's incoming_only_ member seems unused and there was a TODO
about this. This CL just removes it.
Bug: webrtc:10198
Change-Id: I216c291159a32fa2924309affa3769a4be116fd0
Reviewed-on: https://webrtc-review.googlesource.com/c/120931
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26538}
