Commit Graph

28 Commits

Author SHA1 Message Date
f42cc9d8d9 Add MakeUnique from chromium and change StunMessage::AddAttribute to take a unique_ptr.
BUG=NONE

Review-Url: https://codereview.webrtc.org/2757893003
Cr-Commit-Position: refs/heads/master@{#17403}
2017-03-27 23:17:19 +00:00
f137e97adb Revert of Removing HTTPS and SOCKS proxy server code. (patchset #2 id:20001 of https://codereview.webrtc.org/2731673002/ )
Also needed to revert these CLs, which removed code used by the
code being un-removed:
https://codereview.webrtc.org/2745523004
https://codereview.webrtc.org/2754033003
https://codereview.webrtc.org/2758943002

Reason for revert:
This code is still being used by native application developers, so we should send a PSA announcing the deprecation and suggest an alternative before removing it.

Original issue's description:
> Removing HTTPS and SOCKS proxy server code.
>
> This isn't used any more so there's no point in maintaining it.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2731673002
> Cr-Commit-Position: refs/heads/master@{#17016}
> Committed: a1991c5175

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2766063005
Cr-Commit-Position: refs/heads/master@{#17369}
2017-03-23 22:45:49 +00:00
a1991c5175 Removing HTTPS and SOCKS proxy server code.
This isn't used any more so there's no point in maintaining it.

BUG=None

Review-Url: https://codereview.webrtc.org/2731673002
Cr-Commit-Position: refs/heads/master@{#17016}
2017-03-03 18:23:08 +00:00
26d99c2e28 Add the URL attribute to cricket::Candiate.
The URL of the ICE server will be reconstructed by the Port and the URL
attribute is added to the cricket::Candidate struct so that we can tell
which ICE server the candidate was gathered from.

This CL only changes the native C++ code. The Java and Objc wrapper will
be created in separate CLs.

BUG=webrtc::7128

Review-Url: https://codereview.webrtc.org/2685053004
Cr-Commit-Position: refs/heads/master@{#16591}
2017-02-13 20:47:27 +00:00
cc99bc25d8 Change StunMessage::AddAttribute return type from bool to void.
Proper error handling was missing, using VERIFY to crash in debug
builds, while release builds would ignore the error and leak the
attribute memory. The check of attribute type consistency was changed
to a RTC_DCHECK.

Also removes a large number of uses of the deprecated VERIFY macro.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2665343002
Cr-Commit-Position: refs/heads/master@{#16413}
2017-02-02 09:31:30 +00:00
ede5da4960 Replace ASSERT by RTC_DCHECK in all non-test code.
Bulk of the changes were produced using

  git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
    xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'

followed by additional includes of base/checks.h in affected files,
and git cl format.

Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.

Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
2017-01-12 13:15:36 +00:00
0483362377 Add disabled certificate check support to IceServer PeerConnection API.
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".

Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.

PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.

TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.

Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.

For security reasons we ensure that:

	1) The policy is always explicitly initialized to secure.
        2) API users have to explicitly integrate with the feature to
           use it, and will otherwise get no change in behavior.
	3) The feature is not immediately exposed in non-native
	   contexts. For example, disabling of certificate validation
           is not implemented via URI parsing since this would
           immediately allow it to be used from a web page.

This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/

BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
2017-01-09 16:35:45 +00:00
d5236e2948 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ )
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.

Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 	1) The policy is always explicitly initialized to secure.
>         2) API users have to explicitly integrate with the feature to
>            use it, and will otherwise get no change in behavior.
> 	3) The feature is not immediately exposed in non-native
> 	   contexts. For example, disabling of certificate validation
>            is not implemented via URI parsing since this would
>            immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9e

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
2016-12-20 10:22:06 +00:00
b0f04fdb9e Add disabled certificate check support to IceServer PeerConnection API.
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS.

Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.

PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.

TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.

Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.

For security reasons we ensure that:

	1) The policy is always explicitly initialized to secure.
        2) API users have to explicitly integrate with the feature to
           use it, and will otherwise get no change in behavior.
	3) The feature is not immediately exposed in non-native
	   contexts. For example, disabling of certificate validation
           is not implemented via URI parsing since this would
           immediately allow it to be used from a web page.

BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
2016-12-19 12:10:30 +00:00
277b250936 Refactor "secure bool" into explicit PROTO_TLS.
BUG=none

Review-Url: https://codereview.webrtc.org/2568833002
Cr-Commit-Position: refs/heads/master@{#15572}
2016-12-13 13:17:31 +00:00
c309e0e3ea Don't stop sending media on EWOULDBLOCK
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.

We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.

I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2192963002 .

Cr-Commit-Position: refs/heads/master@{#13566}
2016-07-29 00:15:30 +00:00
5d97a9a05b Adding more detail to MessageQueue::Dispatch logging.
Every message will now be traced with the location from which it was
posted, including function name, file and line number.

This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).

This logging should help us identify messages that are taking
longer than expected to be dispatched.

R=pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2019423006 .

Cr-Commit-Position: refs/heads/master@{#13104}
2016-06-10 21:17:33 +00:00
36f50e8e4e Create a new connection if a candidate reuses an address
If the remote side sends a candidate with the same address and port with an existing candidate,
but with a new ufrag and pwd, the local client will create a new connection from it
and destroy the old connection with the same remote address.

BUG=webrtc:5915

Review-Url: https://codereview.webrtc.org/2018693002
Cr-Commit-Position: refs/heads/master@{#13000}
2016-06-01 22:57:12 +00:00
1bffc1d1a4 Rename rtc::Time64 --> rtc::TimeMillis.
In the discussion on https://codereview.webrtc.org/1888593004/, a more
decriptive name was suggested for Time64.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/1923213002
Cr-Commit-Position: refs/heads/master@{#12594}
2016-05-02 15:19:00 +00:00
f1f87203d7 Split ByteBuffer into writer/reader objects.
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.

BUG=webrtc:5155,webrtc:5670

Review URL: https://codereview.webrtc.org/1821083002

Cr-Commit-Position: refs/heads/master@{#12160}
2016-03-30 13:43:44 +00:00
34b11eb66e Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.

The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.

BUG=webrtc:5636

Review URL: https://codereview.webrtc.org/1793553002

Cr-Commit-Position: refs/heads/master@{#12019}
2016-03-16 15:55:48 +00:00
55674ffb32 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}
2016-01-14 14:49:23 +00:00
e5e0e57bdf 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}
2016-01-14 12:57:03 +00:00
7307952a5b 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}
2016-01-14 12:15:56 +00:00
c1aeaf0dc3 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}
2015-10-15 14:26:17 +00:00
0c4e06b4c6 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}
2015-10-07 10:23:32 +00:00
3d564c1015 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}
2015-08-19 23:51:22 +00:00
1cf6f8101a 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}
2015-05-15 17:40:34 +00:00
ff689be3c0 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
2015-02-12 11:55:32 +00:00
332331fb01 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
2014-11-06 20:19:22 +00:00
269fb4bc90 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
2014-10-28 22:20:11 +00:00
28100cb388 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
2014-10-17 22:03:39 +00:00
d1ba6d9cbf 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
2014-10-15 17:30:28 +00:00