438 Commits

Author SHA1 Message Date
Tommi
7a0bdb602c Update PeerConnectionSdpMethods::AddRemoteCandidate
...to use string_view for the mid and prefer .mid() over .name for
ContentInfo.

Bug: webrtc:42233761
Change-Id: Ia9bfe1d7454759ff87295939cda6a71e53cb6b98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374663
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43781}
2025-01-21 08:24:22 -08:00
Philipp Hancke
bce57cda1e Piggyback DTLS handshake in initial STUN packets
This change puts the DTLS handshake as payload of STUN packets with a custom STUN attribute (registered with the IANA) and starts the DTLS handshake before the ICE transport becomes writable. Effectively, STUN acts as a transport layer for DTLS during the handshake phase.

This will theoretically reduce the call setup time by one RTT for aggressive nomination or two RTTs for regular nomination.

The latest DTLS packet (flight) is cached and sent on every STUN request or response. DTLS packets are extracted from every authenticated STUN request or response and handled to the DTLS layer for processing.
The caching also increases the resilience to packet loss as STUN pacing is more aggressive (every 20ms) than the exponential backoff used by DTLS which should reduce call setup time in lossy networks.

If the other side of the connection does not support this feature the fallback to normal DTLS happens as soon as the ICE transport becomes writable. This also handles edge-cases like fragmentation of the DTLS handshake.

The feature is only supported when ECDSA certificates are used since RSA certificates are too large to transport as STUN attributes. The observed attributes for the server and client flights with the certificates were around 600 to 650 bytes. This may be further reduced by using raw public keys defined in RFC 7250.

This feature is disabled by default and guarded by the field trial
  WebRTC-IceHandshakeDtls
and requires experimentation and standardization before roll-out in the browser.

Parts of this landed in
  https://webrtc-review.googlesource.com/c/src/+/370679

BUG=webrtc:367395350

Change-Id: I4809438b2a267c4690a9b2bd6f1766d2f959500d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362480
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43742}
2025-01-15 07:54:23 -08:00
Philipp Hancke
cfaba8fd2d Measure SDP munging
by storing
  [[LastCreatedOffer]] / [[LastCreatedAnswer]]
which are similar to the W3C equivalent but as
description objects instead of serialized SDP strings.

While rejecting all SDP munging is not feasible, this lets us
measure and reject certain modifications gradually.

Chromium metrics CL:
  https://chromium-review.googlesource.com/c/chromium/src/+/6089633

This is measured at three points during the lifetime of a peerconnection:
* for the first SLD call
* when the connection is first established
* when the connection was established and is being closed

Note that the "first" SDP munging detected is returned which may hide that something uses more than one modification.

BUG=chromium:40567530

Change-Id: I964e3ee6e75f73b777d90556fac8691a6f3dc27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43741}
2025-01-15 07:38:45 -08:00
Boris Tsirkin
825379f4dc Format /pc folder
Formatting done via:

git ls-files | grep -E '^pc\/.*\.(h|cc|mm)' | xargs clang-format -i

No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: I3d04503bab53c12927bf408dc63b92cde545b4c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373900
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43689}
2025-01-08 11:55:45 -08:00
Philipp Hancke
740d726739 Move DTLS related code from p2p/base to p2p/dtls
BUG=webrtc:367395350

Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-10 15:55:26 +00:00
Philipp Hancke
1cf342a321 Add IceConfig getter to IceTransportInternal(Interface)
and misc cleanup

BUG=webrtc:367395350
No-Iwyu: remaining IWYU failure is deep inside gtest which is unrelated to the changes and needs to be investigated separately

Change-Id: I5c2b7a6cc6b15fc5474c55eb98635cb9145b7373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370180
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43498}
2024-12-05 09:13:32 +00:00
Harald Alvestrand
fb62f90706 Verify that transport-cc is used when RFC8888 field trial is off.
This is preparatory to ensuring that transport-cc gets turned off when
RFC8888 ccfb is negotiated.

Bug: webrtc:378698658
Change-Id: Ie76677bd6aa046701562bbd93d8489858488f863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368543
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43426}
2024-11-19 13:27:58 +00:00
Harald Alvestrand
0c6d31919e Enable RFC 8888 feedback if negotiated
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.

This should eliminate the need for the "force" flag in the field trial.

Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-14 06:27:45 +00:00
Emil Vardar
d3562286a0 Do not change crypto options in peer_connection.cc
Bug: webrtc:358039777
Change-Id: Icae795a122e0113c64fabd69d0fc2222e9562765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365360
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43245}
2024-10-16 08:18:27 +00:00
Emil Vardar
81d5ab8efb Add field trial to enable negotiation of encrypted RTP header extensions
Bug: webrtc:358039777
Change-Id: I2c59f4c8a3ed16862077c7c3484c1b5b39864c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364661
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#43203}
2024-10-09 09:34:33 +00:00
Olov Brändström
4baeed3b97 Use environment monotonic timestamps (i.e. not UTC) in RTCStats.
Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.

Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
2024-10-03 09:07:17 +00:00
Harald Alvestrand
d153de6d33 Add payload type assignment to offer/answer generation.
This adds payload types to the codecs at the time when offer
is being generated, if they are unassigned at that point.

Bug: webrtc:360058654
Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-17 12:31:25 +00:00
Jeremy Leconte
83d1f9abd0 Ensure <sys/socket.h> is included by using "rtc_base/net_helpers.h".
* IWYU export <sys/socket.h> from rtc_base/net_helpers.h.
* Add a presubmit check to ensures that <sys/socket.h> is included through net_helpers.h (expect if there is a IWYU pragma or a no-presubmit-check).
* Clean up existing includes of <sys/socket.h>

Change-Id: I4bc6cce045c046287f8f74f89edfc9321293b274
Bug: b/236227627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362082
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42996}
2024-09-10 14:23:24 +00:00
Harald Alvestrand
dc56a36ff8 Use PayloadTypePicker in WebRtcVoiceEngine
This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code.

Bug: webrtc:360058654
Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-09 18:44:21 +00:00
Florent Castelli
c5b9a609ea Propagate environment to RtpSenders
Will be later used to conditionally enable mixed codec simulcast
with a field trial.

Bug: webrtc:42220378
Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42929}
2024-09-03 11:56:22 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Harald Alvestrand
5308652c73 Reland "Add recording of PT->Codec mappings on setting SDP for transport"
This reverts commit 6793f831ffdc598e12aced80a4d97956ca50e436.

Reason for revert: Removed the check that caused the error.

Original change's description:
> Revert "Add recording of PT->Codec mappings on setting SDP for transport"
>
> This reverts commit 15717236c8621cb684bb7753acfedbf34d931c80.
>
> Reason for revert: pr-answer
>
> Original change's description:
> > Add recording of PT->Codec mappings on setting SDP for transport
> >
> > Bug: webrtc:360058654
> > Change-Id: I2aa5e0058346cd3fcda47a8ea5115848fbc4f3e2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360041
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42819}
>
> Bug: webrtc:360058654
> Change-Id: I1fea51b3a0cecfa7e7de75f94f47a85fa064be59
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360380
> Reviewed-by: Jonas Oreland <jonaso@google.com>
> Commit-Queue: Jonas Oreland <jonaso@google.com>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42835}

Bug: webrtc:360058654
Change-Id: I2b60ccd60df3bacbeecd848c3cb86f6725b1505a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42847}
2024-08-26 11:11:43 +00:00
Jonas Oreland
6793f831ff Revert "Add recording of PT->Codec mappings on setting SDP for transport"
This reverts commit 15717236c8621cb684bb7753acfedbf34d931c80.

Reason for revert: pr-answer

Original change's description:
> Add recording of PT->Codec mappings on setting SDP for transport
>
> Bug: webrtc:360058654
> Change-Id: I2aa5e0058346cd3fcda47a8ea5115848fbc4f3e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360041
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42819}

Bug: webrtc:360058654
Change-Id: I1fea51b3a0cecfa7e7de75f94f47a85fa064be59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360380
Reviewed-by: Jonas Oreland <jonaso@google.com>
Commit-Queue: Jonas Oreland <jonaso@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42835}
2024-08-23 08:56:51 +00:00
Harald Alvestrand
15717236c8 Add recording of PT->Codec mappings on setting SDP for transport
Bug: webrtc:360058654
Change-Id: I2aa5e0058346cd3fcda47a8ea5115848fbc4f3e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360041
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42819}
2024-08-21 09:06:51 +00:00
Philipp Hancke
4158678b46 Split "helpers" from SSL target to "crypto_random" and rename
since it contains helpers mostly related to cryptographically secure random numbers and strings.

BUG=webrtc:339300437

Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-07 06:41:51 +00:00
Harald Alvestrand
d78e30e00b Deprecate cricket::VideoCodec and cricket::AudioCodec
These are aliases for cricket::Codec.
Also remove internal usage

Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
2024-04-29 16:24:51 +00:00
Victor Boivie
cd54fd8606 sctp: Pass webrtc::Environment to DcSctpTransport
The DcSctpTransport will soon use field trials to conditionally enable
some options.

And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.

Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
2024-03-08 09:45:12 +00:00
Harald Alvestrand
974044efca Remove code for supporting SDES
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}
2024-02-08 14:34:04 +00:00
Per K
39ac25d6ec Add PeerConnectionInterface::ReconfigureBandwidthEstimation
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".


Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.

Integration test in https://webrtc-review.googlesource.com/c/src/+/337322

Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
2024-02-07 14:10:02 +00:00
Harald Alvestrand
8c371f2a9b Reland "Take out Fuchsia-only SDES-enabling parameters"
This is a reland of commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892

Landing after taking out the Chrome usages.

Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}

Bug: webrtc:11066, chromium:804275
Change-Id: I31414dfb6a0ecfa7b6fd91c68603cfd6146869d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337260
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41660}
2024-02-02 17:02:29 +00:00
Olga Sharonova
c0741e9f12 Revert "Take out Fuchsia-only SDES-enabling parameters"
This reverts commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892.

Broke WebRTC into Chrome rolls:

https://chromium-review.googlesource.com/c/chromium/src/+/5248171?tab=checks

/../third_party/blink/renderer/modules/peerconnection/rtc_peer_connection_handler.cc:216:18: error: no member named 'enable_dtls_srtp' in 'webrtc::PeerConnectionInterface::RTCConfiguration'
  216 |   configuration->enable_dtls_srtp = dtls_srtp_key_agreement;
      |   ~~~~~~~~~~~~~  ^

Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}

Bug: webrtc:11066, chromium:804275
Change-Id: I2c2114873091e0c662977a6ef5723e6447166a65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337181
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41643}
2024-01-31 14:35:19 +00:00
Harald Alvestrand
59f3b35013 Take out Fuchsia-only SDES-enabling parameters
This does not remove all traces of SDES - we still need to delete
the cricket::CryptoParams struct and all code that uses it.

Bug: webrtc:11066, chromium:804275
Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41634}
2024-01-30 10:50:12 +00:00
Tommi
698b4e7087 Update more Candidate type checkers to use Candidate::is_*
This is a follow up to a previous CL that removed direct dependency on
the `cricket::` string globals.

Bug: none
Change-Id: I4d839a36739fc4694ce81b72ee036e83dae580df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41623}
2024-01-26 13:41:09 +00:00
Tommi
3b2b2afdaa Move candidate types from port to candidate.h
Add is_* getters to check candidate type without using the string constants directly.

Bug: none
Change-Id: I82c83c032a30a1c67de2d5d6168ecc04e0254318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41568}
2024-01-19 09:24:37 +00:00
Judith Hemp
e56055220b Remove expired histograms WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings
Bug: chromium:1508060
Change-Id: I4a66e53d0c59c320e1ca3cb5a7afa3caf1275064
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331840
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Judith Hemp <hempjudith@google.com>
Cr-Commit-Position: refs/heads/main@{#41412}
2023-12-19 09:12:18 +00:00
Tommi
d6601ce66b Remove PeerConnection::GetRtpTransport
This function isn't used anymore.

Bug: webrtc:9987
Change-Id: I37f1c86cc4802950347db302e8a9207b9dd370bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330261
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41327}
2023-12-06 15:52:37 +00:00
Harald Alvestrand
24510d43dc Delete deprecated AsyncResolver and related classes
To be submitted after downstream usage has been removed, but no earlier than December 1, 2023.

Bug: webrtc:12598
Change-Id: Id9acbac591c48c0c5883fe8f06cf6a68471b70f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323004
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41290}
2023-11-30 15:36:55 +00:00
Danil Chapovalov
3bdb49b483 Create PeerConnection specific environment
Bug: webrtc:15656
Change-Id: I11616e3470798b43cb07a776f5d58669d629e24d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41283}
2023-11-30 09:54:24 +00:00
Danil Chapovalov
49c35d377b In PeerConnection postpone RtcEventLog destruction
This is done as a preparation to move RtcEventLog ownership into Environment where destruction happens later, when all users of the Environment are deleted.

Bug: webrtc:15656
Change-Id: I2a72c74f1fabb1e25c5200aa47a5d61e4b3d9cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41272}
2023-11-29 11:18:31 +00:00
Tomas Gunnarsson
3a15ba6fbf Reland^2 "Reland: Remove unsupported configuration value, allow_codec_switching"
This reverts commit 117d847901ea231cd86ca152b359b88619b9de20.

Reason for revert: Downstream error has been corrected.

Original change's description:
> Revert "Reland: Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.
>
> Reason for revert: Breaks downstream features
>
> Original change's description:
> > Reland: Remove unsupported configuration value, `allow_codec_switching`
> >
> > This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
> >
> > Reason for revert: Relanding once downstream issues have been addressed
> >
> > Original change's description:
> > > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> > >
> > > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
> > >
> > > Reason for revert: breaks downstream
> > >
> > > Original change's description:
> > > > Remove unsupported configuration value, `allow_codec_switching`
> > > >
> > > > Bug: webrtc:11341
> > > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#40995}
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40998}
> >
> > Bug: webrtc:11341
> > Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41032}
>
> Bug: webrtc:11341
> Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41161}

Bug: webrtc:11341
Change-Id: I4a5390a3b8c5e665b742fc564709847ad8853ba9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328160
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41213}
2023-11-22 13:22:08 +00:00
Philipp Hancke
0967247662 Measure usage of fingerprints with SHA-1 certificates
at time of connect. This may allow deprecating SHA-1 which
is no longer used by browsers and not supported by the JS
API.

BUG=None

Change-Id: Iae1d800a61d46e0dcdb622ccb009acc6fb7db53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327540
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41187}
2023-11-17 16:26:17 +00:00
Tomas Gunnarsson
117d847901 Revert "Reland: Remove unsupported configuration value, allow_codec_switching"
This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.

Reason for revert: Breaks downstream features

Original change's description:
> Reland: Remove unsupported configuration value, `allow_codec_switching`
>
> This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
>
> Reason for revert: Relanding once downstream issues have been addressed
>
> Original change's description:
> > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> >
> > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
> >
> > Reason for revert: breaks downstream
> >
> > Original change's description:
> > > Remove unsupported configuration value, `allow_codec_switching`
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40995}
> >
> > Bug: webrtc:11341
> > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40998}
>
> Bug: webrtc:11341
> Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41032}

Bug: webrtc:11341
Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41161}
2023-11-15 08:10:28 +00:00
Harald Alvestrand
a6544377bc Remove not-needed webrtc:: prefixes in pc/
This test drives the new tools_webrtc/remove_extra_namespace.py tool.

Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
2023-11-13 13:23:04 +00:00
Tomas Gunnarsson
23501a2aa6 Reland: Remove unsupported configuration value, allow_codec_switching
This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.

Reason for revert: Relanding once downstream issues have been addressed

Original change's description:
> Revert "Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
>
> Reason for revert: breaks downstream
>
> Original change's description:
> > Remove unsupported configuration value, `allow_codec_switching`
> >
> > Bug: webrtc:11341
> > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40995}
>
> Bug: webrtc:11341
> Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40998}

Bug: webrtc:11341
Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41032}
2023-10-28 16:07:41 +00:00
Tommi
af27d4ea38 Initialize worker_thread_safety_ without BlockingCall().
Bug: webrtc:15099
Change-Id: Iac448c768fb90154fbe5b64fb12d68398a314e9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324281
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41009}
2023-10-25 23:00:52 +00:00
Philip Eliasson
6b0c5babe0 Revert "Remove unsupported configuration value, allow_codec_switching"
This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.

Reason for revert: breaks downstream

Original change's description:
> Remove unsupported configuration value, `allow_codec_switching`
>
> Bug: webrtc:11341
> Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40995}

Bug: webrtc:11341
Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40998}
2023-10-24 08:19:46 +00:00
Tommi
8f7a17f80f Remove unsupported configuration value, allow_codec_switching
Bug: webrtc:11341
Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40995}
2023-10-24 05:07:25 +00:00
Tommi
5b186e98bc Remove effectively dead code for allow_codec_switching
Bug: webrtc:11341
Change-Id: I88e3c1059f5ebcc9d693c0719534aaacd4b9199b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324283
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40990}
2023-10-23 14:08:11 +00:00
Tommi
aea49c953c Simplify PeerConnection::SetConfiguration
* Consolidate ice candidate pool size checks (was in 3 places)
* Consolidate ICE server configuration parsing (was in 2 locations)
* Remove separate blocking call in PC for SetActiveResetSrtpParams().
* Remove unnecessary blocking call inside SetActiveResetSrtpParams
  implementation.

Bug: none
Change-Id: I38c8964f82f91c77c1fd18c407aefaab1d0c7c0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40984}
2023-10-22 15:13:54 +00:00
Tommi
2919075ce3 Remove an invoke for datahannel transport uninitialization during Close.
Bug: none
Change-Id: Ic0d482a8a045d3aa0fcaf13e43f8a156fa3560d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40982}
2023-10-21 16:39:05 +00:00
Tommi
840cf78600 Move Destroy/Create steps for DataChannelTransport to PeerConnection.
This moves steps from the sdp code for pc state over to the PC class
and slightly simplifies the contract between the two classes.
Moving forward it's easier to consolidate those steps in the PC
class with other grouped operations e.g. during teardown.

Also removing GetDataMid() method in favor of the sctp_mid() property.

Bug: none
Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40981}
2023-10-21 16:25:11 +00:00
Philipp Hancke
36e4dd2f42 Add histogram for DTLS peer signature algorithm
in order to estimate the impact of deprecating SHA1. Chromium UMA CL:
  https://chromium-review.googlesource.com/c/chromium/src/+/4894345

BUG=webrtc:15517

Change-Id: I5216ba2a8cbba2f276af20d31aa5e111e7c3a141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321620
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40882}
2023-10-06 12:25:37 +00:00
Harald Alvestrand
4d25a77fd3 Deprecate AsyncResolver config fields and remove internal usage.
Bug: webrtc:12598
Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40627}
2023-08-25 14:02:27 +00:00
Florent Castelli
43a5dd86c2 Implement codec selection api
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.

Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
2023-08-24 13:18:04 +00:00
Harald Alvestrand
b8617d14a6 Use the AsyncDnsResolver in PeerConnection defaults
Bug: webrtc:12598
Change-Id: I1be306e4dbb7c85aa1ccf0fabe96c8556fd5af42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317441
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40613}
2023-08-23 20:29:55 +00:00