When the initial offer side uses the ICE lite implementation, and
initiates a peer connection with an endpoint with the full
implementation, the offer side assumes the controlled ICE role per
RFC5245 and the remote endpoint MUST take the controlling role.
This logic was partially implemented in SetRemoteTransportDescription in
reflection where the endpoint switches its role to the controlling after
receiving the offer. The bug was caused by the following
SetLocalDescription at the remote endpoint after creating the answer,
which overrides the role to the controlled since it has no initial offer
and the role is not reflected in SetLocalTransportDescription. This
results in no nomination of candidate pairs and timeout of establishing
the peer connection.
The fix adds reflection on one's ICE role in SetLocalTransportDescription.
This fix also takes into account the case when both sides use the lite
implementation of ICE and the initial offer side MUST take the controlling
role per RFC5245 in this case, which is the default behavior in the
current implementation.
Bug: webrtc:8531
Change-Id: I65edd296c155bff51fcdb28709975e6837f302d5
Reviewed-on: https://webrtc-review.googlesource.com/26780
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21053}
|packet_overhead| field is added to rtc::NetworkRoute structure.
In PackTransportInternal:
1. network_route() is added which returns the current network route.
2. debug_name() is removed.
3. transport_name() is moved from DtlsTransportInternal and
IceTransportInternal to PacketTransportInternal.
When the selected candidate pair is changed, the P2PTransportChannel
will fire the SignalNetworkRouteChanged instead of
SignalSelectedCandidatePairChanged to upper layers.
The Rtp/SrtpTransport takes the responsibility of calculating the
transport overhead from the BaseChannel so that the BaseChannel
doesn't need to depend on P2P layer transports.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: If9928b25a7259544c2d9c42048b53ab24292fc67
Reviewed-on: https://webrtc-review.googlesource.com/22767
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20664}
This reverts commit 71677452f9cf210aa98162c6f4bd8d339e625337.
Reason for revert: Broke Chromium.
Original change's description:
> Replaced the SignalSelectedCandidatePairChanged with a new signal.
>
> |transport overhead| field is added to rtc::NetworkRoute structure.
>
> In PackTransportInternal:
> 1. network_route() is added which returns the current network route.
> 2. debug_name() is removed.
> 3. transport_name() is moved from DtlsTransportInternal and
> IceTransportInternal to PacketTransportInternal.
>
> When the selected candidate pair is changed, the P2PTransportChannel
> will fire the SignalNetworkRouteChanged instead of
> SignalSelectedCandidatePairChanged to upper layers.
>
> The Rtp/SrtpTransport takes the responsibility of calculating the
> transport overhead from the BaseChannel so that the BaseChannel
> doesn't need to depend on P2P layer transports.
>
> Bug: webrtc:7013
> Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c
> Reviewed-on: https://webrtc-review.googlesource.com/13520
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20661}
TBR=steveanton@webrtc.org,zhihuang@webrtc.org,pthatcher@webrtc.org
Change-Id: Ie0c76786855b65bb8caba7065593c961e4bf9de7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7013
Reviewed-on: https://webrtc-review.googlesource.com/22764
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20662}
|transport overhead| field is added to rtc::NetworkRoute structure.
In PackTransportInternal:
1. network_route() is added which returns the current network route.
2. debug_name() is removed.
3. transport_name() is moved from DtlsTransportInternal and
IceTransportInternal to PacketTransportInternal.
When the selected candidate pair is changed, the P2PTransportChannel
will fire the SignalNetworkRouteChanged instead of
SignalSelectedCandidatePairChanged to upper layers.
The Rtp/SrtpTransport takes the responsibility of calculating the
transport overhead from the BaseChannel so that the BaseChannel
doesn't need to depend on P2P layer transports.
Bug: webrtc:7013
Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c
Reviewed-on: https://webrtc-review.googlesource.com/13520
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20661}
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.
Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
AddRef() now returns void, and Release() returns an enum
RefCountReleaseStatus, to indicate whether or not this Release
call implied deletion.
Bug: webrtc:8270
Change-Id: If2fb77f26118b61751b51c856af187c72112c630
Reviewed-on: https://webrtc-review.googlesource.com/3320
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20366}
The TransportController was in p2p/base before and it cannot depend on
pc/ or media/ level targets because of the circular dependency. To make the
TransportController be responsible for creating and managing
the RtpTransport related objects which are pc/ level targets, the
TransportController is moved from p2p/base to pc/.
The TransportController makes more sense in pc/ anyway, since its main
responsibility is processing the "transport" parts of SDP which is
PeerConnection-specific.
This is also easier than moving RtpTransport related objects to p2p/base
because those objects also depend on other media/ and pc/ level targets
such as srtpfilter, cryptoparams etc.
Bug: webrtc:7013
Change-Id: Ic48dd5c454046ff3c81331f4b459f96a3255f328
Reviewed-on: https://webrtc-review.googlesource.com/4560
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20049}
