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.h
|
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.h
|
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.h
|
381b4217cb36f434c56e399a852a0a15522a9596 |
|
04-Dec-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Ping backup connection at a slower rate and make it configurable from the app. Changed the decision on whether a connection is pingable: 1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection. 2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate. Note the default behavior is the same as before. Also cached the channel state since we are accessing it more often. BUG=webrtc:5034 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1455033004 . Cr-Commit-Position: refs/heads/master@{#10900}
/external/webrtc/webrtc/p2p/base/port.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
8937437872d8ff5b30866ae4de4abf005afc7539 |
|
24-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Do not prune if the current best connection is weak. Otherwise, we may delete a useful connection because the current best connection may be failing. BUG= Review URL: https://codereview.webrtc.org/1364683002 Cr-Commit-Position: refs/heads/master@{#10063}
/external/webrtc/webrtc/p2p/base/port.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
0ba1533fdbe4a098723da8262f1374d71c3a1806 |
|
10-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Added support for an Origin header in STUN messages. For WebRTC there are instances where it may be desirable to provide information to the STUN/TURN server about the website that initiated a peer connection. This modification allows an origin string to be included in the MediaConstraints object provided by the browser, which is then passed as a STUN header in communications with the server. A separate change will be submitted to the Chromium project that uses and is dependent on this change, implementing IETF draft http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02 Originally a patch from skobalt@gmail.com. (https://webrtc-codereview.appspot.com/12839005/edit) R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/port.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|
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.h
|