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}
Calling VirtualSocketServer::SetSendingBlocked(true) will simulate the
network interface being blocked, and SetSendingBlocked(false) will
simulate it being unblocked, resulting in SignalReadyToSend if
appropriate.
I plan to use this to write tests for upper layers of code that deal
with EWOULDBLOCK/SignalReadyToSend.
Also doing some minor housekeeping in this CL (using RTC_DCHECK,
renaming variables, etc.).
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2284903002 .
Cr-Commit-Position: refs/heads/master@{#14170}
This helps a lot on Android devices where the user threads can be scheduled with low priority when the app is in the background, causing spurious significantly delayed before a packet can be read from the socket. With this patch the timestamp is taken by the kernel when the packet actually arrives.
R=juberti@chromium.orgTBR=juberti@webrtc.org
BUG=webrtc:5773
Review URL: https://codereview.webrtc.org/1944683002 .
Cr-Commit-Position: refs/heads/master@{#12850}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
BUG=chromium:468375 (in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1316363005
Cr-Commit-Position: refs/heads/master@{#9913}
Bug 4865: even without STUN/TURN, as long as the peer is on the open internet, the connectivity should work. This is actually a regression even for hangouts.
We need to issue the 0.0.0.0 candidate into Port::candidates_ and filter it out later. The reason is that when we create connection, we need a local candidate to match the remote candidate.
The same connection later will be updated with the prflx local candidate once the STUN ping response is received.
BUG=webrtc:4865
R=juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1274013002 .
Cr-Commit-Position: refs/heads/master@{#9708}
UDP case should not be changed.
Active TCPConnection will initiate Reconnect after OnClose and when Send or Ping fails.
Passive TCPConnection will prune itself as usual as the active side will create a new connection.
The Reconnect could make P2PCT choose a different best_connection in the case where connectivities exist b/w more than 1 Network.
Also, to avoid upper layer triggers ice restart, the WRITE_TIMEOUT caused by the socket disconnection is delayed to give the reconnect mechanism chance to kick in. The timeout event is only fired if the reconnect can't work in 5 sec. If the reconnect, there should be no ICE disconnected state trigger either in active or passive side.
BUG=1926
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31359004
Cr-Commit-Position: refs/heads/master@{#8929}
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.
Not inlining virtual functions with simple bodies such as
{ return false; }
strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.
BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47429004
Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
