and use uint8_t instead of unsigned char. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/365274
BUG=webrtc:357776213
Change-Id: Ibc97e5cc85316ba69b4133b7f3c42e3afbdd7abd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365540
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43263}
This is a reland of commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d
with more backward compat which also fixes the off-by-one issue which caused wrong SRTP keys to be extracted.
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I5d43dc23f90ef630834fb400751979fcc5e18203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43225}
This reverts commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d.
Reason for revert: breaks downstream compilation
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I03ffcda3d6821718f355b243ce78a9c54b4036f3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365062
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43202}
and simplify the interface used as this is only used for exporting
SRTP keys and passing arcane OpenSSL arguments around does not make
much sense.
BUG=webrtc:357776213
Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43198}
Move it away from the "proprietary" SSL_CIPHER_get_id and looking up the cipher based on that towards SSL_CIPHER_standard_name.
SSL_CIPHER_get_id and the associated GetSslCipherSuite API is kept around for
WebRTC.PeerConnection.SslCipherSuite.*
UMA metrics and metrics compability (despite not yielding the IANA ids it promises).
BUG=None
Change-Id: Iaa357e3e31dc90abea688cf6ca10c0b40582ef38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363202
Reviewed-by: David Benjamin <davidben@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43097}
since we do not need two adapters with TLS and DTLS modes.
SSLAdapter is the TLS adapter,
SSLStreamAdapter is the DTLS adapter.
BUG=webrtc:353750117
Change-Id: I223917c71c88437339380e1f196dcf3c0e2021c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42713}
In its stead, there's now a SetEventCallback() method.
Bug: webrtc:11943
Change-Id: If936d6e1e23e8a584f06feb123ecf2d450ea4145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319040
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42187}
This is a step towards removing StreamInterface::SignalEvent.
Downstream dependency will need to be updated to call FireEvent()
before further changes can land in webrtc.
Bug: webrtc:11943
Change-Id: Ia7d3f1c43fda52b7cf5bfa082aef3f462553cd67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347884
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42143}
This restricts the interface such that only a single onclose handler
can be set and that only one OnClose() notification will be fired.
That behavior is the same as how the previous sigslot was being
used, but the difference is that, in addition to removing sigslot,
this pattern is now more explicitly checked in the design.
Bug: webrtc:11943
Change-Id: I469c8cab3d62544988c8145b326af60b06b76d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343340
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41920}
Two new socket options are introduced OPT_SEND_ECN used for setting ECN bits. OPT_RECV_ECN used for reading the ECN bits.
If ECN bits are set on received IP packets, ECT(1) and CE is propagated via rtc::ReceivedPacket.
Bug: webrtc:15368
Change-Id: I3ac335007e2f7d30564569bbc80ce47fa541bef1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332380
Reviewed-by: Jonas Oreland <jonaso@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41885}
Remove internal use of SignalSSLHandshakeError and prepare removal of
sigslot dependency from SSLStreamAdapter.
Bug: webrtc:11943
Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40695}
showing whether the DTLS transport is active and will send
a client hello or not.
BUG=None
Change-Id: If5c631c5e8701b0d1a4bb484467c93027b994ed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287863
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38886}
Updates the code to use the new interfaces
Bug: webrtc:14632
Change-Id: I33b2a25b5968de0251e3cbc84076afc013ecef6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282680
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38601}
This is a reland of commit 02b5f3c9c12cddf3fc6e9125238b77ddb44f3b53
without making SetRemoteFingerprint private (but adding a deprecation warning)
Original change's description:
> dtls: allow dtls role to change during DTLS restart
>
> which is characterized by a change in remote fingerprint and
> causes a new DTLS handshake. This allows renegotiating the
> client/server role as well.
> Spec guidance is provided by
> https://www.rfc-editor.org/rfc/rfc5763#section-6.6
>
> BUG=webrtc:5768
>
> Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#37821}
Bug: webrtc:5768
Change-Id: I8dd674db8b683160013e1b4aa7776775d130978f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37838}
This reverts commit 02b5f3c9c12cddf3fc6e9125238b77ddb44f3b53.
Reason for revert: SetRemoteFingerprint called by downstream code.
Original change's description:
> dtls: allow dtls role to change during DTLS restart
>
> which is characterized by a change in remote fingerprint and
> causes a new DTLS handshake. This allows renegotiating the
> client/server role as well.
> Spec guidance is provided by
> https://www.rfc-editor.org/rfc/rfc5763#section-6.6
>
> BUG=webrtc:5768
>
> Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#37821}
Bug: webrtc:5768
Change-Id: I266b7fdc9cc0b6dc9d3fa732fca37407b98e0816
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272220
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37822}
which is characterized by a change in remote fingerprint and
causes a new DTLS handshake. This allows renegotiating the
client/server role as well.
Spec guidance is provided by
https://www.rfc-editor.org/rfc/rfc5763#section-6.6
BUG=webrtc:5768
Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37821}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
There is a suspicion that it causes OpenSSL errors:
[openssl_stream_adapter.cc(961)]
OpenSSLStreamAdapter::Error(SSL_write, 5, 0)
This commit does change the interaction with OpenSSL as propagating the
socket write errors as SR_BLOCK results in calling BIO_set_retry_write,
as part of current implementation of OpenSSLStreamAdapter.
Testing this regression has proven to be hard to do manually.
This reverts commit edfaaef086ccff2dbff29d64c9a8d9f633637c57.
Bug: webrtc:12943
Change-Id: Ib6767bd4af68c59fd3b7cb051341876f175bb921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230420
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34875}
The UDP sockets in WebRTC are non-blocking, and when writing too much
to them so that their send buffer becomes exhausted, they will return
EAGAIN or EWOULDBLOCK, which indicates that the client will need to
retry a bit later.
Media packets are generally sent by the pacer, which generally avoids
this exhaustion, but for SCTP which has a congestion control algorithm
quite similar to TCP, it may overshoot the amount of data it writes. If
the SCTP library can be notified when writing fails, it can stop writing
for a while until the socket recovers, which will result in less
overshooting and fewer lost packets (possibly even none). But for the
SCTP library to be able to know this, errors must be propagated, which
they weren't with the argument that packets may get lost anyway.
Bug: webrtc:12943
Change-Id: I9244580abf0d48ff810da30a23f995d12be623ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228439
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34751}
Make a few more members const, remove members that aren't used,
set max ssl version number on construction and remove setter.
Bug: none
Change-Id: I6c1a7cabf1e795e027f1bc53b994517e9aef0e93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33622}
- This contains a CallbackList disconnect and handled it
by taking the given subscription tag to subscribe and unsubscribe.
- Left the original sigslot variable until downstream is update after
this change.
Bug: webrtc:11943
No-Try: True
Change-Id: Ie96d74b9594eae11beaa552f61e40f451242bfab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203780
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33285}
- Usage of these sigslots are removed in previous changes in WebRTC
and downstream repositories.
- Remove one more usage of the variables in port_unnittests.
No-Try: True
Bug: webrtc:11943
Change-Id: Ia424f598248a5d9a0cf88f041641a3dd8aa6effe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206500
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33205}
- Signal is transferred from jsep_transport_controller to dtls_transport,
jsep_transport_controller is already using Callbacklist and this
modified the dtls_transport to use callback_list.
Bug: webrtc:11943
Change-Id: I4a7ed08e6dab21b8eb515d4d8971f9b084fb8c86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203722
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33137}
This is a reland of 40261c3663fe316cfe40262c59cee993165ccf63
Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
added a new member with a different name and used it in webrtc code.
After this change do two more follow up CLs to completely remove the old code
from google3.
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
Bug: webrtc:11943
Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32359}
This reverts commit 40261c3663fe316cfe40262c59cee993165ccf63.
Reason for revert: Breaks downstream project
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org
Change-Id: Icf438f87c3d95940d858db3cc5848b23abb82fc4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32324}
- Replace all the top level signals from jsep_transport_controller.
- There are still sigslot usages in this file so keep the inheritance
and that is the reason for not having a binary size gain in this CL.
Bug: webrtc:11943
Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32321}
Also remove test code that can cause leaks into production.
Add sequence checkers.
Bug: webrtc:11988
Change-Id: I67b4cec6ee77d73ccffbbc88c9081ebb3c3cc423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185503
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32228}
It's not expected this will make a difference, since the packet should
be read from the queue if possible as soon as it's added to it.
But we're doing this as an added precaution in case we overlooked
something. See linked bug.
Bug: chromium:1063834
Change-Id: I7a3a6d86a97683cbcbeed5ef1aaa8090cf6bf8c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172661
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30990}
We thought we had resolved this issue earlier, by reading DTLS
records in a loop. But this condition may be triggered in other cases,
such as when an internal DTLS error occurs and more DTLS records
continue to be received afterwords.
Changing this from a hard to soft error will avoid a crash (which
is happening more frequently for whatever reason) and hopefully
enable us to collect logs to debug the issue further.
Bug: chromium:1063834
Change-Id: I22c01a9e064a9db65bab38d00c62a424b5a27437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30955}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
This CL propagates a "closed" signal from DTLS up to the
SCTP section of the data channel controller, where it causes
closing of all open datachannels.
Bug: chromium:1030631, webrtc:10360
Change-Id: I88bb9e1aff5c25f330edfd092ef609d4fcc3a9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162206
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30099}
Bug: chromium:1018077
Change-Id: I585d4064f39e5f9d268b408ebf6ae13a056c778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158403
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29628}
There were two different codepaths that could trigger identical LOGs.
b/136184428
Bug: None
Change-Id: I3297c4e957177c3ffdd4c120cfa1b17d250f0a47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155582
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29381}
