This reverts commit 815522782a.
Reason for revert: Breaks a downstream project.
The internal investigation is still in-progress.
Original change's description:
> sdp: add rtcp-fb:* lines for common feedback
>
> which potentially allows switching to that pattern in the future.
> Video FEC mechanisms (ulpfec, flexfec-03, RED) that currently
> do not have any feedback parameters but will still be considered "common" and feedback may be sent for them.
>
> For audio this causes rtcp-feedback to be sent for G711 and G722 if negotiated.
>
> BUG=webrtc:14802
>
> Change-Id: I54852d39e176f918d4b36462526ceb40617b8fbe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290702
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39224}
Bug: webrtc:14802
Change-Id: I4dc3c0c53ad1bc06050c0d73b088303312ac58b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293020
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39296}
In current state, the SDP parser in webrtc is not backward compatible with clients that might still be using RTP data channels.
Obviously, this isn't there is no such usecase in webrtc since the code is deleted, but in Meta we still use it and would like
to be able to negotiate between clients that offer RTP data channels.
Instead of erroring the parsing procedure, we can parse it as unsupported media in the client that no longer supports RTP data channels.
Replaced the existing test that expects parsing failures with a test that validates that the content was parsed as unsupported media.
Bug: webrtc:14872
Change-Id: I4c105cf55e33b8c19b2849e16148b8175053c40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291190
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39252}
which potentially allows switching to that pattern in the future.
Video FEC mechanisms (ulpfec, flexfec-03, RED) that currently
do not have any feedback parameters but will still be considered "common" and feedback may be sent for them.
For audio this causes rtcp-feedback to be sent for G711 and G722 if negotiated.
BUG=webrtc:14802
Change-Id: I54852d39e176f918d4b36462526ceb40617b8fbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290702
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39224}
preparing to put them at session level when max-bundle is set.
Drive-by: move m= serialization to helper.
BUG=None
Change-Id: I04d918ee8eb70c0cc40baf8ebc12054c6b3a2a15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288820
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38950}
https://www.rfc-editor.org/rfc/rfc8830.html#section-3.2.2
says
Check if a MediaStream with the same WebIDL "id"
attribute already exists. If not, create it.
Ignoring duplicates here satisfies this and brings the behavior
closer to Firefox:
https://github.com/w3c/webrtc-pc/issues/2803
Also make tests use a std::string for the sdp input string.
BUG=webrtc:14745
Change-Id: Iccaabc08d865b779416f6ba4d2dfd5cff04133f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286422
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38840}
parse
a=msid:<stream_id>
since JSEP stipulates sending this syntax as track identifers
have become meaningless. The track id will be set to a random string.
a=msid:<stream_id> <track_id>
remains supported for backward compability.
BUG=webrtc:14729
Change-Id: I86c073eb97cd613324271125de18a773235fc79d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38814}
The DCHECK crashes debug builds running some applications such as Webex.
Bug: None
Change-Id: I0061286c4c1d04964678a00014896f1fccd4685d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276460
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38644}
to allow for downstream users to upgrade.
BUG=chromium:1338902
Change-Id: Ie1205ad2c9c1be3f4ed8e133b1a5e54afd04ebd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268193
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37501}
to allow for downstream users to upgrade.
BUG=chromium:1338902
Change-Id: If6b56ab63f7859c13e9ebc70326e1088e5dfff1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268141
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37475}
RID is defined for multiple usages in RFC 8851, but we only support
usage with a=simulcast as specified in RFC 8853.
Bug: chromium:1341043
Change-Id: Ie72074c5b394bdc41865938a86ec9c7629e1f5e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267628
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37417}
This is a reland of commit ad6807805d
Original change's description:
> sdp: reject duplicate codecs with the same id but different name or clockrate
>
> since something like
> rtpmap:96 VP8/90000
> rtpmap:96 VP9/90000
> or
> rtpmap:97 ISAC/32000
> rtpmap:97 ISAC/16000
> is wrong. Note that fmtp or rtcp-fb are not taken into account.
> Also note that sending invalid static payload types now throws an error.
>
> Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
>
> BUG=None
>
> Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#37028}
Bug: webrtc:14140
Change-Id: I63a37aacea6b9e0a9d7570b8422849275eb69aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264544
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37066}
This reverts commit ad6807805d.
Reason for revert: Speculative revert due to consistent Mac browser
test failures preventing WebRTC from rolling int Chromium:
https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/10410/overview
"Failed to parse SessionDescription. a=rtpmap:103 ISAC/16000 Duplicate payload type with conflicting codec name, clock rate or number of channels."
Original change's description:
> sdp: reject duplicate codecs with the same id but different name or clockrate
>
> since something like
> rtpmap:96 VP8/90000
> rtpmap:96 VP9/90000
> or
> rtpmap:97 ISAC/32000
> rtpmap:97 ISAC/16000
> is wrong. Note that fmtp or rtcp-fb are not taken into account.
> Also note that sending invalid static payload types now throws an error.
>
> Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
>
> BUG=None
>
> Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#37028}
Bug: None
Change-Id: Ic9c06c9309bb68bd94bfdd2e30ffd6ff96f6812b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37064}
since something like
rtpmap:96 VP8/90000
rtpmap:96 VP9/90000
or
rtpmap:97 ISAC/32000
rtpmap:97 ISAC/16000
is wrong. Note that fmtp or rtcp-fb are not taken into account.
Also note that sending invalid static payload types now throws an error.
Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
BUG=None
Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37028}
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}
the maximum used in practice is multiopus with
6 or 8 channels. 24 is the maximum number of channels
supported in the audio decoder.
BUG=chromium:1265806
Change-Id: Iba8e3185a1f235b846fed9c154e66fb3983664ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/main@{#35440}
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}
Change types of const std::string& arguments.
Use absl::string_view for the reference to input, to prepare for
parsing with less copies. Use std::string (passed by value) for the
description, to support ownership transfer without copying.
Bug: None
Change-Id: I4358b42bb824e4eb7a5ac9b64d44db1b9b022bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223667
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34721}
This reverts commit 37ee0f5e59.
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}
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.
This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).
PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.
C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/
Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.
Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
regression from
https://webrtc-review.googlesource.com/c/src/+/197813
which attempted to cast the unsupported content type with
a sctp protocol to a application/datachannel one.
BUG=chromium:1171965
Change-Id: I87c63da83b9f49d968e9b045bb1079f687ab226e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33100}
The extmap-allow-mixed SDP attribute signals that one- and two-byte RTP
header extensions can be mixed. In practice, this also means that WebRTC
will support two-byte RTP header extensions when this is signaled by
both peers.
Bug: webrtc:9985
Change-Id: I80a3f97bab162c7d9a5acf2cae07b977641c039d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197943
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33036}
for style consistency. This check is already done outside the method.
BUG=None
Change-Id: Ie1366fa57417258a301b02503ad76f304f4279a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32853}
and reorganise the parsing
Bug: None
Change-Id: I21f08297429a0cc0265da00daa681d934fc43d66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196643
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32843}
otherwise this shows up in the logs as unhandled when it has been handled.
BUG=None
Change-Id: Ic081312a266d7a7ffff6220d2979cefa29a8591e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196652
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32810}
The a=rtcp:9 IN IP4 0.0.0.0 line is required by JSEP to be generated,
but is also required to be ignored. This reduces log spew.
Bug: None
Change-Id: I984060d9693b9df4c4cfdf2c5dea0ea620f4bc83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196641
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32798}
Example of current output in appr.tc:
https://paste.googleplex.com/4582802164023296
No-Try: True
Bug: None
Change-Id: I9b717b9c13e771e84682d9e3d3ee6b0920a85a44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196526
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32796}
the limit is ignored anyway. Also rename rtp datachannel
bandwidth limit constant.
BUG=webrtc:6625
Change-Id: If7b26691ced8148955e98c86b9bed692b2e55e8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189972
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32479}
This is a reland of 239f92ecf7
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32457}
