Commit Graph

388 Commits

Author SHA1 Message Date
376e1235c7 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}
2015-11-25 17:00:12 +00:00
953eabc027 Revert "GetDefaultLocalAddress should return false when the address is invalid"
This reverts commit 67c6df6153b7b6dceb2b569daf683a498b2fc13c.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1470363002 .

Cr-Commit-Position: refs/heads/master@{#10780}
2015-11-24 20:00:38 +00:00
67c6df6153 GetDefaultLocalAddress should return false when the address is invalid
BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1471203002 .

Cr-Commit-Position: refs/heads/master@{#10779}
2015-11-24 19:59:26 +00:00
521ed7bf02 Reland Convert internal representation of Srtp cryptos from string to int
TBR=pthatcher@webrtc.org
BUG=webrtc:5043

Review URL: https://codereview.webrtc.org/1458023002 .

Cr-Commit-Position: refs/heads/master@{#10703}
2015-11-19 03:42:00 +00:00
318166bed7 Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
Reason for revert:
Broke chromium fyi build.

Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}

TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043

Review URL: https://codereview.webrtc.org/1455233005

Cr-Commit-Position: refs/heads/master@{#10702}
2015-11-19 03:03:46 +00:00
2764e1027a Convert internal representation of Srtp cryptos from string to int.
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.

External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.

The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().

BUG=webrtc:5043

Review URL: https://codereview.webrtc.org/1416673006

Cr-Commit-Position: refs/heads/master@{#10701}
2015-11-19 02:02:40 +00:00
310b093aa4 Fix active tcp port to 9
In tcp only call:
Tested with hangout.
Tested with firefox.

To test firefox, goto about:config, search for media.peerconnection.ice.tcp and turn it on.

Existing test case should be suffice to cover this.

R=juberti@google.com
TBR=jubert@webrtc.org
BUG=webrtc:3849

Review URL: https://codereview.webrtc.org/1217463004 .

Cr-Commit-Position: refs/heads/master@{#10683}
2015-11-18 03:15:57 +00:00
32f39968ce Re-apply change https://codereview.webrtc.org/1426673007/
Do not delete the turn port entry right away when the respective
connection is deleted. The dependency on asyncinvoker has been added
in chromium libjingle-nacl.

BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1450263002

Cr-Commit-Position: refs/heads/master@{#10679}
2015-11-17 19:36:37 +00:00
54e92326af Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ )
Reason for revert:
I have to revert this unfortunately because it adds a dependency on AsyncInvoker, which is not included when building libjingle_nacl in Chromium.
AsyncInvoker needs to first be added to the list of sources in Chromium.

Original issue's description:
> Do not delete the turn port entry right away when the respective connection is deleted.
> BUG=webrtc:5120
>
> Committed: https://crrev.com/e58fe8ef0e6d959f54adee3ed77764927d3845cc
> Cr-Commit-Position: refs/heads/master@{#10641}

TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1449863002

Cr-Commit-Position: refs/heads/master@{#10649}
2015-11-16 12:13:02 +00:00
e58fe8ef0e Do not delete the turn port entry right away when the respective connection is deleted.
BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1426673007

Cr-Commit-Position: refs/heads/master@{#10641}
2015-11-14 01:54:47 +00:00
fa5d0dbd1e cleanup: get rid of basicdefs.h include
The ARRAY_SIZE macro it defines is not used anymore, as all the usages
were converted to arraysize macro from arraysize.h.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1443273002

Cr-Commit-Position: refs/heads/master@{#10640}
2015-11-13 22:37:43 +00:00
a4845efe14 Fix flaky tests
BUG=webrtc:5190,webrtc:5136

Review URL: https://codereview.webrtc.org/1405103006

Cr-Commit-Position: refs/heads/master@{#10639}
2015-11-13 17:53:01 +00:00
2cd7afe7e2 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}
2015-11-12 19:14:38 +00:00
8597543ae8 Schedule a CreatePermissionRequest after the success of a previous request
unless a channel binding request is already scheduled.

BUG=5178
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1434603002 .

Cr-Commit-Position: refs/heads/master@{#10625}
2015-11-12 19:07:25 +00:00
a8e9f5e4e4 A little cleanup in p2ptransportchannel and transportchannel.
No functional changes.

BUG=

Review URL: https://codereview.webrtc.org/1411103012

Cr-Commit-Position: refs/heads/master@{#10614}
2015-11-12 00:15:12 +00:00
9b5ee9c0d9 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}
2015-11-11 21:19:25 +00:00
653b8e02f2 Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
Reason for revert:
Relanding with compile warning fixed.

Original issue's description:
> Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
>
> Reason for revert:
> Caused compiler warning, breaking Chrome FYI bots.
>
> Original issue's description:
> > Adding the ability to change ICE servers through SetConfiguration.
> >
> > Added a SetIceServers method to PortAllocator. Also added a new
> > PeerConnection Initialize method that takes a PortAllocator, in the
> > hope that we can get rid of PortAllocatorFactoryInterface, since the
> > only substantial thing a factory does is convert the webrtc:: ICE
> > servers to cricket:: versions.
> >
> > Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> > Cr-Commit-Position: refs/heads/master@{#10420}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/18a944bf0ac9eed872dc009bd58e6bc12c946303
> Cr-Commit-Position: refs/heads/master@{#10421}

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1414313003

Cr-Commit-Position: refs/heads/master@{#10609}
2015-11-11 20:55:18 +00:00
e03cab94c1 When running this code in chromium on a machine with IPv6 disabled, the RTC_DCHECK fails and in release build, it could leak to further crash in chromium's rtc_peer_connection_hanlder.cc.
Here is the right fix.

BUG=webrtc:5061
R=pthatcher@google.com
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1437933002 .

Cr-Commit-Position: refs/heads/master@{#10607}
2015-11-11 19:11:28 +00:00
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
ad13d2f817 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}
2015-11-11 00:34:58 +00:00
9af97f8910 WebRTC should generate default private address even when adapter enumeration is disabled.
Introduce a DefaultAddressProvider such that rtc::Network can't access other part of NetworkManager.

This also removes the hack of generating the loopback address. The dependency has been removed by https://codereview.chromium.org/1417023003/

BUG=webrtc:5061
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1411253008 .

Cr-Commit-Position: refs/heads/master@{#10590}
2015-11-10 22:47:49 +00:00
9b66957b4f Stop a session when a new connection becomes writable.
We cannot do it at the end of sorting because it may stop a session too early.
Also remove was_writable_, which is not useful.
BUG=webrtc:5119

Review URL: https://codereview.webrtc.org/1406423008

Cr-Commit-Position: refs/heads/master@{#10511}
2015-11-04 20:07:49 +00:00
20a3461908 Remove deprecated IsUnresolved() method from SocketAddress API.
This patch removes IsUnresolved() method and update the clients to use
IsUnresolvedIP() instead.

BUG=None
R=perkj@webrtc.org
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1414793006

Cr-Commit-Position: refs/heads/master@{#10487}
2015-11-03 00:20:28 +00:00
8a4f547dad Hang on android when DNS resolution is not done
BUG=webrtc:5139
R=juberti@google.com
TBR=juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1429493009 .

Cr-Commit-Position: refs/heads/master@{#10463}
2015-10-30 16:12:45 +00:00
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
77d0d6e858 When all connections timed out on writing, delete them all.
BUG=5111

Review URL: https://codereview.webrtc.org/1421123003

Cr-Commit-Position: refs/heads/master@{#10431}
2015-10-27 18:34:50 +00:00
18a944bf0a Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
Reason for revert:
Caused compiler warning, breaking Chrome FYI bots.

Original issue's description:
> Adding the ability to change ICE servers through SetConfiguration.
>
> Added a SetIceServers method to PortAllocator. Also added a new
> PeerConnection Initialize method that takes a PortAllocator, in the
> hope that we can get rid of PortAllocatorFactoryInterface, since the
> only substantial thing a factory does is convert the webrtc:: ICE
> servers to cricket:: versions.
>
> Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> Cr-Commit-Position: refs/heads/master@{#10420}

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1424803004

Cr-Commit-Position: refs/heads/master@{#10421}
2015-10-27 02:21:45 +00:00
d3b26d9439 Adding the ability to change ICE servers through SetConfiguration.
Added a SetIceServers method to PortAllocator. Also added a new
PeerConnection Initialize method that takes a PortAllocator, in the
hope that we can get rid of PortAllocatorFactoryInterface, since the
only substantial thing a factory does is convert the webrtc:: ICE
servers to cricket:: versions.

Review URL: https://codereview.webrtc.org/1391013007

Cr-Commit-Position: refs/heads/master@{#10420}
2015-10-27 00:55:27 +00:00
2b5586774c Exposing DTLS transport state from TransportChannel.
This is necessary in order to support the RTCPeerConnectionState enum in
the future, as well as a correct RTCIceConnectionState (which isn't a
combination ICE and DTLS state).

Review URL: https://codereview.webrtc.org/1414363002

Cr-Commit-Position: refs/heads/master@{#10419}
2015-10-27 00:23:34 +00:00
b0bb77fd61 Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1416773003/ )
Reason for revert:
https://codereview.chromium.org/1419253002 is landed to address this linker issue. Keep my fingers crossed.

Original issue's description:
> Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
>
> Reason for revert:
> This CL breaks Chromium, undefined reference link error to webrtc::field_trial::FindFullName. Adding the dependency system_wrappers to the rtc_p2p target is not enough to fix this.
>
> Looking at field_trial.h (in system_wrappers/interface/, not to be confused with the one in test/) the documentation says "WebRTC clients MUST provide an implementation of: ...FindFullName... Or link with a default one provided in: ...system_wrappers.gyp:field_trial_default).
>
> So maybe just depend on field_trial_default? Not sure what should be done in case there are implications to adding this dependency, I'm reverting this. Sorry :)
>
> Original issue's description:
> > Add experiment on weak ping delay during call set up time
> >
> > BUG=
> > R=pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > Cr-Commit-Position: refs/heads/master@{#10343}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>

TBR=pthatcher@webrtc.org,juberti@webrtc.org,hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1413603005

Cr-Commit-Position: refs/heads/master@{#10418}
2015-10-26 22:10:06 +00:00
5d9b92b53d Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
The existing comment is wrong, and the test even ensures it: Bind will capture reference values by reference. That makes it hard to use with AsyncInvoker, because you can't safely Bind to a function that takes (const) reference params.

The new version of this code strips references in the bound object, so it captures by value, but can bind against functions that take const references, they'll just be references to the copy.

As the class comment implies, actual by-reference args should be passed as pointers or things that safely share (e.g. scoped_refptr) and not references directly. A new test case ensures the pointer reference works. The new code will also give a compiler error if you try to bind
to a non-const reference.

BUG=

Review URL: https://codereview.webrtc.org/1291543006

Cr-Commit-Position: refs/heads/master@{#10397}
2015-10-24 18:14:52 +00:00
4f6a8b5f55 Revert of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1406153005/ )
Reason for revert:
Still cause break on mac. reverting it again.

Original issue's description:
> Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1423443002/ )
>
> Reason for revert:
> This should be safe to land now.
>
> Original issue's description:
> > Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
> >
> > Reason for revert:
> > guoweis - Here's the target that's failing:
> > https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle_nacl.gyp&l=17
> >
> > This has unfortunately been causing problems repeatedly for us since libjingle_nacl is maintained separately from libjingle (I don't know the history).
> >
> > The way this works for Chrome in general is that the FindFullName method is implemented in init_webrtc.cc in the overrides folder in Chrome and that hooks WebRTC up with Chrome's implementation.  I'm not sure if that's the right thing to do for nacl, how webrtc is initialized there etc.  I'll ping the nacl team for some help too offline and include you.  Reverting this change for now.
> >
> > Original issue's description:
> > > Add experiment on weak ping delay during call set up time
> > >
> > > BUG=
> > > R=pthatcher@webrtc.org
> > >
> > > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > > Cr-Commit-Position: refs/heads/master@{#10343}
> >
> > TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=
> >
> > Committed: https://crrev.com/a01d44022355796d4fd86d00aae6d3263573b6f1
> > Cr-Commit-Position: refs/heads/master@{#10350}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,tommi@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e26ce1b7a4644942b239ed788a737200762db3b3
> Cr-Commit-Position: refs/heads/master@{#10379}

TBR=pthatcher@webrtc.org,juberti@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1413843003

Cr-Commit-Position: refs/heads/master@{#10380}
2015-10-23 01:00:46 +00:00
e26ce1b7a4 Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1423443002/ )
Reason for revert:
This should be safe to land now.

Original issue's description:
> Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
>
> Reason for revert:
> guoweis - Here's the target that's failing:
> https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle_nacl.gyp&l=17
>
> This has unfortunately been causing problems repeatedly for us since libjingle_nacl is maintained separately from libjingle (I don't know the history).
>
> The way this works for Chrome in general is that the FindFullName method is implemented in init_webrtc.cc in the overrides folder in Chrome and that hooks WebRTC up with Chrome's implementation.  I'm not sure if that's the right thing to do for nacl, how webrtc is initialized there etc.  I'll ping the nacl team for some help too offline and include you.  Reverting this change for now.
>
> Original issue's description:
> > Add experiment on weak ping delay during call set up time
> >
> > BUG=
> > R=pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > Cr-Commit-Position: refs/heads/master@{#10343}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/a01d44022355796d4fd86d00aae6d3263573b6f1
> Cr-Commit-Position: refs/heads/master@{#10350}

TBR=pthatcher@webrtc.org,juberti@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1406153005

Cr-Commit-Position: refs/heads/master@{#10379}
2015-10-23 00:49:34 +00:00
c80741f895 Fixing some issues with the direction attribute of m-lines in offers.
By default, we'll now offer to receive if already receiving
(meaning that the last remote description contained a track).

Also, m-lines that are neither receiving nor sending are now correctly
marked "inactive".

Also moved some logic relating to default tracks out of webrtcsdp.cc,
such that now the direction seen by upper layers will always be
consistent with the consumed/produced SDP.

BUG=528089

Review URL: https://codereview.webrtc.org/1406803004

Cr-Commit-Position: refs/heads/master@{#10376}
2015-10-22 20:14:51 +00:00
a01d440223 Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
Reason for revert:
guoweis - Here's the target that's failing:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle_nacl.gyp&l=17

This has unfortunately been causing problems repeatedly for us since libjingle_nacl is maintained separately from libjingle (I don't know the history).

The way this works for Chrome in general is that the FindFullName method is implemented in init_webrtc.cc in the overrides folder in Chrome and that hooks WebRTC up with Chrome's implementation.  I'm not sure if that's the right thing to do for nacl, how webrtc is initialized there etc.  I'll ping the nacl team for some help too offline and include you.  Reverting this change for now.

Original issue's description:
> Add experiment on weak ping delay during call set up time
>
> BUG=
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> Cr-Commit-Position: refs/heads/master@{#10343}

TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1423443002

Cr-Commit-Position: refs/heads/master@{#10350}
2015-10-21 08:07:32 +00:00
3bf69b15f4 Add experiment on weak ping delay during call set up time
BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1411883002 .

Cr-Commit-Position: refs/heads/master@{#10343}
2015-10-20 19:09:45 +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
d59daf8023 Merging BaseSession code into WebRtcSession.
After the TransportController CL, BaseSession does little more than
hold a state and an error, and act as an intermediary for the
TransportController. So it doesn't make sense for it to be its own
class.

Review URL: https://codereview.webrtc.org/1397973002

Cr-Commit-Position: refs/heads/master@{#10281}
2015-10-14 22:02:50 +00:00
8ac544e811 Get rid of deprecated SocketAddress::IsAny() method.
This patch converts the usage of IsAny() to IsAnyIP() and removes the
deprecated method.

BUG=None
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1392153002

Cr-Commit-Position: refs/heads/master@{#10220}
2015-10-08 14:15:49 +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
6caafbe5b6 Convert uint16_t to int for WebRTC cipher/crypto suite.
This is a follow up CL on https://codereview.webrtc.org/1337673002

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1377733004 .

Cr-Commit-Position: refs/heads/master@{#10175}
2015-10-05 19:43:33 +00:00
456696a9c1 Reland Change WebRTC SslCipher to be exposed as number only
This is to revert the change of https://codereview.webrtc.org/1380603005/

TBR=pthatcher@webrtc.org
BUG=523033

Review URL: https://codereview.webrtc.org/1375543003 .

Cr-Commit-Position: refs/heads/master@{#10126}
2015-10-01 04:49:02 +00:00
27dc29b0df Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
Reason for revert:
This broke chromium.fyi bot.

Original issue's description:
> Change WebRTC SslCipher to be exposed as number only.
>
> This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
>
> For SRTP, currently it's still string internally but is reported as IANA number.
>
> This is used by the ongoing CL https://codereview.chromium.org/1335023002.
>
> BUG=523033
>
> Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943
> Cr-Commit-Position: refs/heads/master@{#10124}

TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=523033

Review URL: https://codereview.webrtc.org/1380603005

Cr-Commit-Position: refs/heads/master@{#10125}
2015-10-01 02:23:15 +00:00
4fe3c9a773 Change WebRTC SslCipher to be exposed as number only.
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.

For SRTP, currently it's still string internally but is reported as IANA number.

This is used by the ongoing CL https://codereview.chromium.org/1335023002.

BUG=523033

Review URL: https://codereview.webrtc.org/1337673002

Cr-Commit-Position: refs/heads/master@{#10124}
2015-10-01 01:49:17 +00:00
d0b3143f0e 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}
2015-09-30 19:42:25 +00:00
c4d3a5d44c Thinning out the Transport class.
Connecting TransportChannelImpls directly to the TransportController,
and removing redundant signal forwarding/state aggregating code from
Transport. This brings us closer to just getting rid of Transport
entirely.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1380563002 .

Cr-Commit-Position: refs/heads/master@{#10120}
2015-09-30 17:33:08 +00:00
2b342bf99c 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}
2015-09-30 16:52:48 +00:00
98db68fdaa If gather_continually is set to true, keep the last port allocator session running while stopping all existing process of getting ports (when p2ptransportchannel first becomes writable).
BUG=5034

Review URL: https://codereview.webrtc.org/1359363003

Cr-Commit-Position: refs/heads/master@{#10110}
2015-09-29 14:58:26 +00:00
1f429e3418 Passing the new policy from PeerConnection RTCConfiguration to
p2ptransportchannel.  This CL does not use the new policy yet.
BUG=

Review URL: https://codereview.webrtc.org/1369773003

Cr-Commit-Position: refs/heads/master@{#10092}
2015-09-28 14:57:39 +00:00
8937437872 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}
2015-09-24 20:14:51 +00:00