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}
This code looked a bit weird before this CL - probably because of old
refactorings.
In JsepTransport constructor, there is a DCHECK assuring that the RTP
DTLS transport is always present, so it can be passed directly to the
SctpTransport constructor, which avoids having the SetDtlsTransport
method in it.
Also, in the SctpTransport constructor, there was code that would set
the SCTP transport state to `kConnecting` if the DTLS transport was
present, but that was dead code, as it was always `nullptr` inside the
constructor before this CL. With this CL, it's always present, and the
SCTP Transport's state will initially always be `kConnecting` now. Which
is a step to deprecating the `kNew` state that doesn't exist in
https://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate.
One test case was modified, as it didn't test the reality. The test
created a SctpTransport, registered an observer, and added the DTLS
transport, and expected to receive a "statechange" from `kNew` (which is
not a state that exists in the spec) to `kConnecting`. If the test had
tested the opposite ordering - adding the DTLS transport first, and then
adding an observer, it wouldn't have experienced this. And since in
reality (with the implementation of JsepTransport before and
after this CL), it always adds the DTLS transport before any observer is
registered. So it wouldn't ever be fired, outside of tests.
Bug: webrtc:15897
Change-Id: I6ac24e0a331b686eb400fcf388ece50f2ad46a32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345420
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41987}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
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}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
Make the jsep_transport.cc code conform to WebRTC code style.
Bug: None
Change-Id: I9142c1e83d4006988e206aeb0bbe52dfffafda60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35631}
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}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.
Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}
TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
Make access to rtcp_transport_ limited to network thread.
Bug: none
Change-Id: Id5c2834f758da595724079596d839e528c92e977
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205982
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33180}
Most members it used to protect or now either const, or accessed on
network thread only.
Followup to https://webrtc-review.googlesource.com/c/src/+/204801.
Bug: webrtc:11567
Change-Id: I1bc80555885a8d8e9f7282d5adf93a093879cc7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205980
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33178}
Also delete the CompositeRtpTransport class, since it is never
instantiated.
Locking intentionally left unchanged in this cl, except for removal of
RTC_GUARDED_BY annotations on the now const members.
Bug: None
Change-Id: I99c22ff528ce7a46f71081b98ca83745b8146afc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205000
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33165}
Also removing need for lock for ice restart flag, fix call paths and
add information about how JsepTransportController's events could live
fully on the network thread and complexity around signaling thread
should be handled by PeerConnection (more details in webrtc:12427).
Bug: webrtc:12426, webrtc:12427
Change-Id: I9b1fae8acf16d90d9716054fc3c390700877a82a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33159}
Mark corresponding webrtc::Mutex as mutable, to allow use in const methods.
Bug: webrtc:11567
Change-Id: Ia8c731a91c719a531799abf24fd30a15b54428af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204801
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33126}
And delete the always null members data_channel_transport_ and
composite_data_channel_transport_ from the JsepTransport class.
Bug: webrtc:9719
Change-Id: Ibfd92b74708d63a75521f6f1d5fbc3830bd67e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179280
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31727}
This patch fixes a problem in https://webrtc.googlesource.com/src/+/71ff07369837d6575c04ebff7002d07d6e0af25f
that when adding standard compliance validation of ufrag/pwd
accidentally broken ice renomination by introducing a new "constructor".
Bug: chromium:1044521
Change-Id: If1b18b1d728e55db9da385b37162a9cb5e61ac48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169549
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30670}
Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport. This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).
This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.
Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
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}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
