We already support decoding of the x-mt line. This change adds the
a=x-mt line to the SDP offer. This is not a backward compatible change
for media transport (because of the changes in pre-shared key handling)
1) if media transport is enabled, and SDES is enabled, generate the
media transport offer.
2) if media transport generated the offer, add that offer to the x-mt
line.
3) in order to create media transport, require an x-mt line (backward incompatible).
The way it works is that
1) PeerConnection, on the offerer, asks jsep transport for the
configuration of the media transport.
2) Tentative media transport is created in JsepTransportController when
that happens.
3) SessionDescription will include configuration from this tentative
media transport.
4) When the LocalDescription is set on the offerer, the tentative media
transport is promoted to the real media transport.
Caveats:
- now we really only support MaxBundle. In the previous implementations,
two media transports were briefly created in some tests, and the second
one was destroyed shortly after instantiation.
- we, for now, enforce SDES. In the future, whether SDES is used will be
refactored out of the peer connection.
In the future (on the callee) we should ignore 'is_media_transport' setting. If
Offer contains x-mt, media transport should be used (if the factory is
present). However, we need to decide how to negotiate media transport
for data channels vs data transport for media (x-mt line at this point
doesn't differentiate the two, so we still need to use app setting).
This change also removes the negotation of pre-shared key from the
a=crypto line. Instead, media transport will have its own, 256bit key.
Such key should be transported in the x-mt line. This makes the code
much simpler, and simplifies the dependency / a=crypto lines parsing.
Also, adds a proper test for the connection re-offer (on both sides: callee and caller).
Before, it was possible that media transport could get recreated, based on the offer.
The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test.
This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even
when there is a re-offer.
Bug: webrtc:9719
Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01
Reviewed-on: https://webrtc-review.googlesource.com/c/125040
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26933}
This change adds a parser for the x-mt line to the WebRTC SDP, and adds a collection in the SessionDescription for media transport.
x-mt line contains information about media transport settings.
This is an experimental line.
This change will be followed up by two other changes:
* Setting the offer with x-mt
* Using the x-mt line to pass it to the media transport
Bug: webrtc:9719
Change-Id: I46568610d4092a69ada7b7c1d3987d698ddba0be
Reviewed-on: https://webrtc-review.googlesource.com/c/124601
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26881}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
In the January 22nd 2019 WebRTC meeting it was agreed that an offer
for sending (or receiving) simulcast should only contain the RIDs
of the layers that are sent by the client.
This change removes the complexity that was added to support sending
and receiving the single layer (and RID) that are sent from the server.
Bug: webrtc:10076
Change-Id: I8bae1336d5cb8ba2f91c5b62332dc69e67ddfd47
Reviewed-on: https://webrtc-review.googlesource.com/c/120242
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26432}
