This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
This is part of "Perfect Negotiation" (https://crbug.com/980872).
Spec PR here (merged): https://github.com/w3c/webrtc-pc/pull/2169
Spec: https://w3c.github.io/webrtc-pc/#dfn-localufragstoreplace
The restartIce() makes the next createOffer() generate new ICE
credentials, as if {iceRestart:true} was passed in as options. It also
causes negotiationneeded. This is better than manually restarting ICE
because it survives rollbacks (when that is implemented) and
restartIce() can be called regardless of current signalingState.
Bug: chromium:980881
Change-Id: I8e70bec31ce9d4d6a303bd35e91b2dcc28fcad60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144941
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28596}
1. Propagate sent notification for all packets, including RTCP
2. Add Field Trial to disable datagram => RTCP feedback loop translation (for tests and experiments only, because having two feedback loops add extra overhead).
Bug: webrtc:9719
Change-Id: Ia3143cc79d127ae331210c86d6675d6e778e962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145460
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28568}
default connection address is a hostname candidate.
Using a FQDN in the c= line has caused an inter-op issue with Firefox
when hostname candidates are the only candidates gathered when forming
the media sections. To address this issue, we use 0.0.0.0:9 when a
hostname candidate would be used to populate the c= and the m= lines.
The SDP grammar related to ICE candidates has been moved out of RFC8445,
and is currently defined in draft-ietf-mmusic-ice-sip-sdp. A FQDN
address must not be used in the connection address attribute per the
latest draft, if the ICE agent generates local candidates. Also, the
wildcard addresses (0.0.0.0 or ::) with port 9 are given the exception
as the connection address that will not result in an ICE mismatch. We
thus adopt the aforementioned solution after combining these
considerations.
Bug: chromium:927309, chromium:982108
Change-Id: I3df2db0f154276da39f99650289cf81baa677e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28547}
This change removes RTCP Feedback loop if we are using datagram transport by removing transport sequence numbers from RTP packets and recreating RTCP Feedback from Datagram ACKs and Timestamps.
- For outgoing RTP packets, remove transport sequence number and store it with datagram_id. Note that removing transport sequence numbers does not only save 4-8 bytes per packet, but also prevents generation of feedback packets on the receiver side.
- When datagram ACKs, we re-created RTCP feedback with timestamp.
- Replacing previous assumption that datagram_id was the same as packet_id by storing incremental counter of datagram ids (I noticed some packets come without packet_id, which is a bit strange, but easy to support and it's also good not to rely on packet_ids being unique across multiple ssrcs).
Bug: webrtc:9719
Change-Id: Iecfe938ecea1a74e7c9e1484f0e985d72643d4a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145269
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28542}
Datagram_dtls_adaptor needs access to rtp_rtcp modules and this moves helps to keep p2p/base/ without dependency on rtp_rtcp.
Bug: webrtc:9719
Change-Id: Ic337be3fb9f68106187a84efa815eefbe5b0fcd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145267
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28533}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
Because of webrtc:10801, we don't actually support 4 simulcast layers but 3.
Until this is fixed, we limit the value to what we can currently handle.
Bug: webrtc:8785
Change-Id: I513b7c8d4c889fa0d80c91adc1c4f874acb86fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144625
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28487}
Implements RTCAudioSourceStats members:
- audioLevel
- totalAudioEnergy
- totalSamplesDuration
In this CL description these are collectively referred to as the audio
levels.
The audio levels are removed from sending "track" stats (in Chrome,
these are now reported as undefined instead of 0).
Background:
For sending tracks, audio levels were always reported as 0 in Chrome
(https://crbug.com/736403), while audio levels were correctly reported
for receiving tracks. This problem affected the standard getStats() but
not the legacy getStats(), blocking some people from migrating. This
was likely not a problem in native third_party/webrtc code because the
delivery of audio frames from device to send-stream uses a different
code path outside of chromium.
A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the
send-side audio levels to the RTCAudioSourceStats, while keeping the
receive-side audio levels on the "track" stats. This allows an
implementation to report the audio levels even if samples are not sent
onto the network (such as if an ICE connection has not been established
yet), reflecting some of the current implementation.
Changes:
1. Audio levels are added to RTCAudioSourceStats. Send-side audio
"track" stats are left undefined. Receive-side audio "track" stats
are not changed in this CL and continue to work.
2. Audio level computation is moved from the AudioState and
AudioTransportImpl to the AudioSendStream. This is because a) the
AudioTransportImpl::RecordedDataIsAvailable() code path is not
exercised in chromium, and b) audio levels should, per-spec, not be
calculated on a per-call basis, for which the AudioState is defined.
3. The audio level computation is now performed in
AudioSendStream::SendAudioData(), a code path used by both native
and chromium code.
4. Comments are added to document behavior of existing code, such as
AudioLevel and AudioSendStream::SendAudioData().
Note:
In this CL, just like before this CL, audio level is only calculated
after an AudioSendStream has been created. This means that before an
O/A negotiation, audio levels are unavailable.
According to spec, if we have an audio source, we should have audio
levels. An immediate solution to this would have been to calculate the
audio level at pc/rtp_sender.cc. The problem is that the
LocalAudioSinkAdapter::OnData() code path, while exercised in chromium,
is not exercised in native code. The issue of calculating audio levels
on a per-source bases rather than on a per-send stream basis is left to
https://crbug.com/webrtc/10771, an existing "media-source" bug.
This CL can be verified manually in Chrome at:
https://codepen.io/anon/pen/vqRGyq
Bug: chromium:736403, webrtc:10771
Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28480}
Section 5.2 of draft-ietf-mmusic-sdp-simulcast-14:
The direction for an rid-id MUST be aligned with the direction
specified for the corresponding RTP stream identifier on
the "a=rid" line.
Bug: webrtc:10785
Change-Id: I1fc70706511ae17c821c5ec4d90a0b854171454f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144245
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28452}
Currently, use_datagram_transport's non-default value is never used.
Instead of reading configuration.use_datagram_transport,
PeerConnection::Initialize reads the local configuration's
use_datagram_transport. This hasn't been set yet, and so it always
falls back to the default value.
Bug: webrtc:9719
Change-Id: I028ed537c7d88ee3421b6bd92fc7d5e3c6970529
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144441
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28451}
Currently, GenerateOrGetLastMediaTransportOffer() creates a media
transport that has an RtcEventLog, regardless of whether the media
transport is used for media or data channels. It should only set the
RtcEventLog when used for media.
Bug: webrtc:10789
Change-Id: Id91c16973deec89bbc8c6518c4c9f1039f1265fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144367
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28450}
First, the existing configuration parameter (use_datagram_transport) is
now optional.
The new field trial has two flag values:
1. Whether to enable the datagram transport (enabled)
2. Whether to use the datagram transport by default (default_value)
The first is a kill-switch. It disables the datagram transport, even
for applications which inject a datagram transport factory and specify
use_datagram_transport = true. This allows applications which hard-code
a datagram transport to switch it off via field trials.
This flag defaults to true, to avoid breaking downstream projects which
already inject and configure a datagram transport. It may be changed to
false after updating downstream to set this field trial flag to true
when required.
The second provides a default value to be used in case the
aforementioned use_datagram_transport parameter is unset. Applications
which explicitly set use_datagram_transport will use that value.
Applications which do not explicitly specify whether or not to use the
datagram transport will use it (or not) according to the default_value
flag.
One goal of this flag is to simplify rollout in applications which
already set field trials based on configuration, but require code
changes for new RTCConfiguration parameters. A second goal is to
provide platforms with a knob to control whether datagram transport is
"opt-in" or "opt-out".
This flag defaults to false, to prevent downstream projects from
unintentionally enabling the datagram tranpsort.
Bug: webrtc:9719
Change-Id: I521a5fa61c992e76e5081118678a1812a261d672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28435}
Only remaining user is WavReader. Demote its constructor
accepting a PlatformFile to private, to refactor implementation
in a later cl.
Bug: webrtc:6463
Change-Id: I7b950be6f02073cb135dd0fab1190b9dc0de1fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144025
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28410}
This CL enables to change surface_ice_candidates_on_ice_transport_type_changed
in RTCConfiguration via PeerConnection::SetConfiguration.
Bug: None
Change-Id: Ib7bc8a08bfc9bf59cf07fe217c6f57d0d63615f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143561
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28394}
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28385}
Per discussions at https://crbug.com/webrtc/10753, the
remote-outbound-rtp.ssrc is supposed to reflect the SSRC of the RTP
media stream (i.e. outbound-rtp.ssrc) and not the sender that the
corresponding RTCP report block was transmitted on.
Bug: webrtc:10753
Change-Id: Id88f5fdbe6397ba81a46f0ef430bd6f08e66b145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143484
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28354}
instead of using factories for MediaEngine and RtcEventLog that rely on GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ie1135f70f4ae4d047c4d6bf2db61489a663385aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141875
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28328}
Users that would like custom TaskQueueFactory (e.g. chromium) should use
CreateModularPeerConnectionFactory directly
Bug: webrtc:10284
Change-Id: I7ba55a0f21560d4beb71a7f93f6fa70b0fe74931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142234
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28324}
Also removes SessionDescription::Copy.
Bug: webrtc:10612
Change-Id: Ib652d717531738c3ed5d1054e32a03961e16dba9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139903
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28239}
He's been working enough in this area to be an owner.
Bug: none
Change-Id: I919d8c6da6d685e5e72c330b2603435f55287561
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140861
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28214}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28198}
This reverts commit 71c6482baf0ff17141c635e6a7639493db68a65c.
Reason for revert: Lands too much at once and breaks downstream tests that need to implement new interfaces first.
Original change's description:
> Implement true negotiation for DatagramTransport with fallback to RTP.
>
> In short, the caller places a x-opaque line in SDP for each m= section that
> uses datagram transport. If the answerer supports datagram transport, it will
> parse this line and create a datagram transport. It will then echo the x-opaque
> line into the answer (to indicate that it accepted use of datagram transport).
>
> If the offer and answer contain exactly the same x-opaque line, both peers will
> use datagram transport. If the x-opaque line is omitted from the answer (or is
> different in the answer) they will fall back to RTP.
>
> Note that a different x-opaque line in the answer means the answerer did not
> understand something in the negotiation proto. Since WebRTC cannot know what
> was misunderstood, or whether it's still possible to use the datagram transport,
> it must fall back to RTP. This may change in the future, possibly by passing
> the answer to the datagram transport, but it's good enough for now.
>
> Negotiation consists of four parts:
> 1. DatagramTransport exposes transport parameters for both client and server
> perspectives. The client just echoes what it received from the server (modulo
> any fields it might not have understood).
>
> 2. SDP adds a x-opaque line for opaque transport parameters. Identical to
> x-mt, but this is specific to datagram transport and goes in each m= section,
> and appears in the answer as well as the offer.
> - This is propagated to Jsep as part of the TransportDescription.
> - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
> media_session.cc, webrtc_sdp.cc
>
> 3. JsepTransport/Controller:
> - Exposes opaque parameters for each mid (m= section). On offerer, this means
> pre-allocating a datagram transport and getting its parameters. On the
> answerer, this means echoing the offerer's parameters.
> - Uses a composite RTP transport to receive from either default RTP or
> datagram transport until both offer and answer arrive.
> - If a provisional answer arrives, sets the composite to send on the
> provisionally selected transport.
> - Once both offer and answer are set, deletes the unneeded transports and
> keeps whichever transport is selected.
>
> 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
>
> Bug: webrtc:9719
> Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28182}
TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org
Change-Id: I0d502c4a6d27516c35ed85154f3fa5869f88b3b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140822
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28188}
This reverts commit 907b5920912dcec98e677621cd65efefb9dc0eab.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Remove another DCHECK that fails during renegotiation.
>
> Also adds a test case that catches both this DCHECK and the previous
> one.
>
> Bug: webrtc:9719
> Change-Id: I590544a13cd178274e9c11b698c4694fd5cf0d59
> No-Try: True
> Tbr: steveanton@webrtc.org
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140802
> Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28185}
TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org
Change-Id: I2f01f720ffe6b9a3d5dd412b607f4e31e8cb1f97
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140823
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28186}
Also adds a test case that catches both this DCHECK and the previous
one.
Bug: webrtc:9719
Change-Id: I590544a13cd178274e9c11b698c4694fd5cf0d59
No-Try: True
Tbr: steveanton@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140802
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28185}
This fires in downstream tests when re-negotiation of a datagram
transport completes.
Bug: webrtc:9719
Change-Id: Ie7337e7dc33e41a83da37d3ef2fda596d9107256
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140821
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28183}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28182}
This is a reland of 9469c784dbf732472e3b2a60a5fcca0a2f432313
Original change's description:
> Added OnIceCandidateError to API and implementation
>
> Bug: webrtc:3098
> Change-Id: I27ffd015ebf9e8130c1288f7331b0e2fdafb01ef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135953
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28173}
TBR=steveanton@webrtc.org
Bug: webrtc:3098
Change-Id: I77af2065fc1479273f399e2b3d919f98fe8ac23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140641
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28179}
Bug: webrtc:3098
Change-Id: I27ffd015ebf9e8130c1288f7331b0e2fdafb01ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135953
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28173}
