| 55674ffb32307c6f3efaab442340d3c5c075073b |
|
14-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.
R=tommi@webrtc.org
TBR=pthatcher@webtrc.org
BUG=4173
Review URL: https://codereview.webrtc.org/1589563003 .
Cr-Commit-Position: refs/heads/master@{#11251}
/external/webrtc/webrtc/p2p/base/port.cc
|
| e5e0e57bdfd8831b2ad917e7990e273fdfe26af4 |
|
14-Jan-2016 |
tommi <tommi@webrtc.org> |
Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
Reason for revert:
Broke Chrome:
https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_chromeos_compile_dbg_ng/builds/143025/steps/compile%20%28with%20patch%29/logs/stdio
FAILED: cd ../../third_party/libjingle; python ../../native_client/build/build_nexe.py --root ../.. --product-dir ../../out/Debug/xyz --config-name Debug -t ../../native_client/toolchain/ --arch pnacl --build newlib_plib --name ../../out/Debug/gen/tc_pnacl_newlib/lib/libjingle_nacl.a --objdir ../../out/Debug/obj/third_party/libjingle/libjingle_nacl.gen/pnacl_newlib-pnacl/libjingle_nacl "--include-dirs=../../out/Debug/gen/tc_pnacl_newlib/include ../.. \"../../out/Debug/gen\" ./source ../ ../../native_client_sdk/src/libraries ../../native_client_sdk/src/libraries/nacl_io/include ../../native_client_sdk/src/libraries/third_party/newlib-extras ../expat/files/lib ../boringssl/src/include" "--compile_flags=-O2 -g -Wall -fdiagnostics-show-option -Werror -Wno-unused-function -Wno-char-subscripts -Wno-c++11-extensions -Wno-unnamed-type-template-args -Wno-extra-semi -Wno-unused-private-field -Wno-char-subscripts -Wno-unused-function \"-std=gnu++11\" " --gomadir /b/build/goma "--defines=\"__STDC_LIMIT_MACROS=1\" \"__STDC_FORMAT_MACROS=1\" \"_GNU_SOURCE=1\" \"_POSIX_C_SOURCE=199506\" \"_XOPEN_SOURCE=600\" \"DYNAMIC_ANNOTATIONS_ENABLED=1\" \"DYNAMIC_ANNOTATIONS_PREFIX=NACL_\" \"NACL_BUILD_ARCH=x86\" V8_DEPRECATION_WARNINGS \"CLD_VERSION=2\" \"_FILE_OFFSET_BITS=64\" CHROMIUM_BUILD \"CR_CLANG_REVISION=255169-1\" COMPONENT_BUILD UI_COMPOSITOR_IMAGE_TRANSPORT \"USE_AURA=1\" \"USE_ASH=1\" \"USE_PANGO=1\" \"USE_CAIRO=1\" \"USE_DEFAULT_RENDER_THEME=1\" \"USE_LIBJPEG_TURBO=1\" \"USE_X11=1\" \"IMAGE_LOADER_EXTENSION=1\" \"ENABLE_WEBRTC=1\" \"ENABLE_MEDIA_ROUTER=1\" USE_PROPRIETARY_CODECS ENABLE_PEPPER_CDMS ENABLE_CONFIGURATION_POLICY ENABLE_NOTIFICATIONS \"ENABLE_HIDPI=1\" \"ENABLE_TOPCHROME_MD=1\" USE_UDEV DONT_EMBED_BUILD_METADATA \"DCHECK_ALWAYS_ON=1\" FIELDTRIAL_TESTING_ENABLED \"ENABLE_TASK_MANAGER=1\" \"ENABLE_EXTENSIONS=1\" \"ENABLE_PDF=1\" \"ENABLE_PLUGINS=1\" \"ENABLE_SESSION_SERVICE=1\" \"ENABLE_THEMES=1\" \"ENABLE_AUTOFILL_DIALOG=1\" \"ENABLE_BACKGROUND=1\" \"ENABLE_PRINTING=1\" \"ENABLE_PRINT_PREVIEW=1\" \"ENABLE_SPELLCHECK=1\" \"ENABLE_CAPTIVE_PORTAL_DETECTION=1\" \"ENABLE_APP_LIST=1\" \"ENABLE_SUPERVISED_USERS=1\" \"ENABLE_MDNS=1\" \"ENABLE_SERVICE_DISCOVERY=1\" V8_USE_EXTERNAL_STARTUP_DATA FULL_SAFE_BROWSING SAFE_BROWSING_CSD SAFE_BROWSING_DB_LOCAL EXPAT_RELATIVE_PATH FEATURE_ENABLE_SSL GTEST_RELATIVE_PATH HAVE_OPENSSL_SSL_H NO_MAIN_THREAD_WRAPPING NO_SOUND_SYSTEM WEBRTC_POSIX SRTP_RELATIVE_PATH SSL_USE_OPENSSL USE_WEBRTC_DEV_BRANCH \"timezone=_timezone\" XML_STATIC \"USE_LIBPCI=1\" \"USE_OPENSSL=1\" \"USE_OPENSSL_CERTS=1\"" "--link_flags=-B../../out/Debug/gen/tc_pnacl_newlib/lib " "--source-list=../../out/gypfiles/third_party/libjingle/pnacl_newlib.libjingle_nacl.source_list.gypcmd"
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:50:23: error: 'CreateConnection' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
virtual Connection* CreateConnection(const Candidate& address,
^
../webrtc/p2p/base/portinterface.h:71:23: note: overridden virtual function is here
virtual Connection* CreateConnection(
^
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:53:16: error: 'PrepareAddress' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
virtual void PrepareAddress();
^
../webrtc/p2p/base/portinterface.h:63:16: note: overridden virtual function is here
virtual void PrepareAddress() = 0;
^
(etc)
Original issue's description:
> Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
>
> To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
>
> BUG=4173
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/7307952a5bf63311e5f9a2a90089a06177e42716
> Cr-Commit-Position: refs/heads/master@{#11247}
TBR=pthatcher@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=4173
Review URL: https://codereview.webrtc.org/1586063002
Cr-Commit-Position: refs/heads/master@{#11249}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 7307952a5bf63311e5f9a2a90089a06177e42716 |
|
14-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
BUG=4173
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1577873003 .
Cr-Commit-Position: refs/heads/master@{#11247}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 37389b42b4f42c49583b5fa187611cdb6dcd67b4 |
|
05-Jan-2016 |
honghaiz <honghaiz@webrtc.org> |
Don't delete an ICE connection until it has been pruned or timed out on writing in the case where it
hasn't received anything yet. Deleting an ICE connection before it is pruned or timed out
when it hasn't received anything yet leads to ICE connections being deleted
before they have a chance to send a ping and receive a response.
BUG=
Review URL: https://codereview.webrtc.org/1544003002
Cr-Commit-Position: refs/heads/master@{#11151}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 376e1235c7b602e86afe9f36eb81289e42643718 |
|
25-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Destroy a Connection if a CreatePermission request fails.
This means that if a TURN server denies permission for an
unreachable address, we'll no longer ping it fruitlessly.
BUG=webrtc:4917
Review URL: https://codereview.webrtc.org/1415313004
Cr-Commit-Position: refs/heads/master@{#10789}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 2cd7afe7e2dc011ab00bdbc131039b16aa8fbdeb |
|
12-Nov-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Do not delete a connection until it has not received anything for 30 seconds.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1422623015 .
Cr-Commit-Position: refs/heads/master@{#10626}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 9b5ee9c0d988b1d0dc64920937448e301dd45bd9 |
|
11-Nov-2015 |
honghaiz <honghaiz@webrtc.org> |
Send back ping response if the ping comes from an unknown address.
BUG=webrtc:5171
Review URL: https://codereview.webrtc.org/1424703012
Cr-Commit-Position: refs/heads/master@{#10610}
/external/webrtc/webrtc/p2p/base/port.cc
|
| ad13d2f8178af5efbe516184995af02a171ec66a |
|
11-Nov-2015 |
Tim Psiaki <tpsiaki@google.com> |
Round Rate computations from RateTracker.
BUG=534221
R=asapersson@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1410533004 .
Cr-Commit-Position: refs/heads/master@{#10592}
/external/webrtc/webrtc/p2p/base/port.cc
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/webrtc/p2p/base/port.cc
|
| d0b3143f0e37f5f5e0578e88cf740dd839b50c24 |
|
30-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Do not time out a port if its role switched from controlled to controlling. Also fix some comments.
BUG=webrtc:5026
Review URL: https://codereview.webrtc.org/1376983002
Cr-Commit-Position: refs/heads/master@{#10122}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 2b342bf99c9578247940929c02a41ef9ccec6d6e |
|
30-Sep-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Delete a connection only if it has timed out on writing and not receiving for 10 seconds.
BUG=webrtc:5034,webrtc:4937
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1371623003 .
Cr-Commit-Position: refs/heads/master@{#10119}
/external/webrtc/webrtc/p2p/base/port.cc
|
| a58ea7806a039c14e1f92a0757123062963b44b1 |
|
24-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any.
This will paves the way for continuous nomination lite and multi-networking.
2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.
Review URL: https://codereview.webrtc.org/1311433009
Cr-Commit-Position: refs/heads/master@{#10057}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 04ac81f2fd8ef6680522438fac1894db5415a0ec |
|
21-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
BUG=4937
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345913004 .
Cr-Commit-Position: refs/heads/master@{#10004}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 275a2f16fd99b0f1eb43fd4ba62541af14e797c0 |
|
21-Sep-2015 |
tommi <tommi@webrtc.org> |
Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ )
Reason for revert:
Broke the Windows build:
[226/365] LINK_EMBED cc_perftests.exe
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj.rsp /c ..\..\remoting\protocol\channel_socket_adapter_unittest.cc /Foobj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj /Fdobj\remoting\remoting_unittests.cc.pdb
e:\b\build\slave\win\build\src\remoting\protocol\channel_socket_adapter_unittest.cc(36) : error C3861: 'set_readable': identifier not found
ninja: build stopped: subcommand failed.
Original issue's description:
> Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
> If a connection does not receive for 30 seconds, it will be deleted.
> BUG=
>
> Committed: https://crrev.com/ae16f8547d3b447f62f6660f13688585c6c3de15
> Cr-Commit-Position: refs/heads/master@{#10001}
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1356103002
Cr-Commit-Position: refs/heads/master@{#10002}
/external/webrtc/webrtc/p2p/base/port.cc
|
| ae16f8547d3b447f62f6660f13688585c6c3de15 |
|
21-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
If a connection does not receive for 30 seconds, it will be deleted.
BUG=
Review URL: https://codereview.webrtc.org/1351673003
Cr-Commit-Position: refs/heads/master@{#10001}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 7cbd188c5ed7df80bb737bd4ada94422730e2d89 |
|
18-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (again).
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1353713002 .
Cr-Commit-Position: refs/heads/master@{#9979}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 6304626268238a074051910d201e9a77aae677e0 |
|
14-Sep-2015 |
Tim Psiaki <tpsiaki@google.com> |
Add a rate tracker that tracks rate over a given interval split up into buckets that accumulate unit counts for their portion of said interval and use this instead of the standard rate tracker so that the values of retrieved frame rate stats are completely independent of the polling rate.
BUG=
R=asapersson@webrtc.org, noahric@chromium.org, pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1279433006 .
Cr-Commit-Position: refs/heads/master@{#9933}
/external/webrtc/webrtc/p2p/base/port.cc
|
| d12140a68efdcffa1c2c18f25149905e9dae1a9c |
|
10-Sep-2015 |
guoweis <guoweis@webrtc.org> |
Revert change which removes GICE.
There are still dependencies on this functionality.
TBR=pthatcher@webrtc.org
BUG=526399
Review URL: https://codereview.webrtc.org/1336553003
Cr-Commit-Position: refs/heads/master@{#9920}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 2159b89fa2cb55beeef38f72bd45e217f3d33d4e |
|
22-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.
Original CL: https://codereview.webrtc.org/1263663002/
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1303393002 .
Cr-Commit-Position: refs/heads/master@{#9761}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 5bdafd44c86ee46bd7e040f19828324583418b33 |
|
21-Aug-2015 |
minyuel <minyue@webrtc.org> |
Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.
Original code review see
https://codereview.webrtc.org/1291363005
The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137
TBR=pthatcher@webrtc.org,
BUG=
Review URL: https://codereview.webrtc.org/1308753003 .
Cr-Commit-Position: refs/heads/master@{#9756}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 5a3acd89648e7cff7e1b76b2da710be041be54a0 |
|
21-Aug-2015 |
honghaiz <honghaiz@webrtc.org> |
First step of passive aggressive nomination.
On the controlled side, a stun request without use-candidate attribute will
be used for sending media.
BUG=4900
Review URL: https://codereview.webrtc.org/1270613006
Cr-Commit-Position: refs/heads/master@{#9747}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 081f34b564e1a26ffbbe9515eba1fef7c736fdde |
|
20-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1291363005 .
Cr-Commit-Position: refs/heads/master@{#9738}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 3d564c10157d7de1d2d4236f4e2a13ff1363d52b |
|
20-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Add instrumentation to track the IceEndpointType.
The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.
BUG=webrtc:4918
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1277263002 .
Cr-Commit-Position: refs/heads/master@{#9737}
/external/webrtc/webrtc/p2p/base/port.cc
|
| fa301809b698017455847f45cc7e0dfa1bdfed35 |
|
11-Aug-2015 |
pthatcher <pthatcher@webrtc.org> |
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 3449faa553ec94c52ef2d0949867befb60992c88 |
|
10-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1263663002 .
Cr-Commit-Position: refs/heads/master@{#9692}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 54360510ff9b7c61fc906d3ed360b06a5824bbf1 |
|
08-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add flakyness check based on the recently received packets.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1207563002 .
Cr-Commit-Position: refs/heads/master@{#9553}
/external/webrtc/webrtc/p2p/base/port.cc
|
| b8b0143a11afd495b8e9c1a1cc388cdbd4340b99 |
|
08-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Tighten link-local routing exclusion check
Also add a unit test for this behavior.
BUG=https://code.google.com/p/webrtc/issues/detail?id=4823
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1218293016 .
Cr-Commit-Position: refs/heads/master@{#9550}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 1fe120a6b9371819515f2f05beaf62ddcc9c9f30 |
|
10-Jun-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add triggered checks.
BUG=4590
R=guoweis@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51979004.
Cr-Commit-Position: refs/heads/master@{#9409}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 42af6caf5c6e67eb33fb5dba9d93f01be0e638b8 |
|
15-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add logging of "use candidate" and when we switch ICE "best" connections.
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46309004
Cr-Commit-Position: refs/heads/master@{#9197}
/external/webrtc/webrtc/p2p/base/port.cc
|
| b2d2623902d9fb676d272ac03710b7cc2d0c5a68 |
|
15-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Don't use rtc::LogCheckLevel, because it breaks Chrome.
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55429004
Cr-Commit-Position: refs/heads/master@{#9196}
/external/webrtc/webrtc/p2p/base/port.cc
|
| 1cf6f8101ae9db517332783e99c98e14ff4c47e1 |
|
15-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add logging for sending and receiving STUN binding requests and TURN requests and responses.
BUG=
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46189004
Cr-Commit-Position: refs/heads/master@{#9195}
/external/webrtc/webrtc/p2p/base/port.cc
|
| be508a1d3634ce63b64cd740c44600453e3c3a6b |
|
06-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Implement Tcp Reconnect for TCPPort.
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}
/external/webrtc/webrtc/p2p/base/port.cc
|
| ff689be3c0c59c1be29aaa0697aa0f762566d6c6 |
|
12-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35079004
Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| dacdd9403d30cdb13ab2de645841edd2ae76950d |
|
23-Jan-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Reland r7980:
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 9657265f391cfe473a61b18a4579bbbeb44c9bd8 |
|
09-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
This reverts r7980.
It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.
Review URL: https://webrtc-codereview.appspot.com/41429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| c5fd66dcdfdba3ec114cc5b5c0337eba503cee40 |
|
29-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Accept incoming pings before remote answer is set to reduce connection latency.
BUG=4068
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 950c51825109c2ca352317edef0a33777d0e6678 |
|
17-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Committed: https://code.google.com/p/webrtc/source/detail?r=7906
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 55360ae402908b24757c7983c587e69ea485e9e6 |
|
16-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Add adapter_type into Candidate object."
This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.
BUG=
TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| aaf02cc2d4f696345ce0e6d5715f2cfa22aea689 |
|
16-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| fb108b5a28a538862a4157e17de795426d86af1e |
|
15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r7885.
Breaks compile step of other code where network name of
cricket::Candidate is used.
TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/31229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 8c9d79a29d9127d4ff8aa4ae386630c72cfb1808 |
|
12-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 8c9ff203c5f3c87891b46067ef6ec21b37d2dde4 |
|
04-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Redo the change of https://webrtc-codereview.appspot.com/30949004/
The previous change causes a build issue as there is subclass of TransportChannel in chromium. To break the circular dependency, a stub of implementation for GetState() is provided and will be removed once the jingle_glue::MockTransportChannel has the function defined.
TBR=pthatcher@webrtc.org
BUG=411086
Review URL: https://webrtc-codereview.appspot.com/34369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7806 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| fd8422938c7031c6bb31e2fa6288d45cbf48cb99 |
|
04-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Implement GetState() for channel's connectivity check state."
This reverts commit ff72f9e692d0918b32646dadaf382aa4355d8437.
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7805 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| ff72f9e692d0918b32646dadaf382aa4355d8437 |
|
04-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Implement GetState() for channel's connectivity check state.
Previously, IceState is considered completed when there is only one connection (and the rest was trimmed). However, since the trimming logic is only done within the scope of network, when IPv6 and IPv4 both exist, the completion event is never fired.
This change adds the GetState() to each channel and it could decide what Completion means. The transport object then aggregates all channels before determining it's completed.
Each channel's IceState will be aggregrated at Transport level for overall Ice state
BUG=411086
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7804 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 930e004a817ed346a99ac8e56575326ca75e72aa |
|
17-Nov-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.
BUG=427555
Copied from https://webrtc-codereview.appspot.com/25959004/
R=harryjin@google.com, juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7693
Review URL: https://webrtc-codereview.appspot.com/32039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 6a782c2a46d83e09bb036d34b8c2363adc26d037 |
|
14-Nov-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
TBR=guoweis@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/25179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 312614a438c2104ccab6d0231d17604359674e15 |
|
13-Nov-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.
BUG=427555
Copied from https://webrtc-codereview.appspot.com/25959004/
R=harryjin@google.com, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 43e033e7785fff6815fb2154cee87af893fe47a4 |
|
10-Nov-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted."
BUG=3379
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7673 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 332331fb01f8a316ac6d61cf4572478610fb3472 |
|
06-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use uint16s for port numbers in webrtc/p2p/base.
This is a necessary precursor to using uint16s for port numbers more
consistently in Chromium code.
This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override).
BUG=chromium:81439
TEST=none
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
|
28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
|
18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|
| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
|
15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.cc
|