The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel.
DtlsSrtpTransport is responsible for exporting keys from DtlsTransport
and setting up the wrapped SrtpTransport.
The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL.
Bug: webrtc:7013
Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713
Reviewed-on: https://webrtc-review.googlesource.com/6700
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20804}
|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 71677452f9.
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}
The existing methods SetEncrypedHeaderExtensionIds in SrtpTransport and SrtpSession
are removed because those methods could be confusing. When these methods are called
the head extension IDs are not actually updated and the user need to call SetRtpParams
again to make that happen. The existing setter just caches the new IDs.
To make it less confusing, the SetEncryptedHeaderExtensionIds is removed and the new
extension IDs will be set immediately when setting the crypto params.
For SDES, the crypto params and the header extension IDs will be set at the same time.
For DTLS, the new header extensions are cached in BaseChannel and will be set when
the DTLS handshake is completed.
Another major change is that when doing DTLS-SRTP, the encrypted header extension
IDs will be updated only when they are changed.
Bug: webrtc:7013
Change-Id: Ib70d4797456ae5ecb61b3dfff15c7e3e7ede89bd
Reviewed-on: https://webrtc-review.googlesource.com/15860
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20639}
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
In SrtpTransport::SetRtcpParams, send_rtcp_session_ should really call
SetSend rather than SetRecv.
Modified the LOG message in SrtpTransport::SetRtpParams.
Bug: webrtc:8436
Change-Id: Iccbfbc5ef2d4f4ebd5f876c3f6dcc81671fdc631
Reviewed-on: https://webrtc-review.googlesource.com/14562
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20403}
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't
know if the keys are from SDES or DTLS handshake.
BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.
SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
BUG=webrtc:7013
Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e
Reviewed-on: https://webrtc-review.googlesource.com/2560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19977}
Reason for revert:
This seems to be causing some video freezes. See https://bugs.chromium.org/p/webrtc/issues/detail?id=8251
Original issue's description:
> Completed the functionalities of SrtpTransport.
>
> The SrtpTransport takes the SRTP responsibilities from the BaseChannel
> and SrtpFilter. SrtpTransport is now responsible for setting the crypto
> keys, protecting and unprotecting the packets. SrtpTransport doesn't know
> if the keys are from SDES or DTLS handshake.
>
> BaseChannel is now only responsible setting the offer/answer for SDES
> or extracting the key from DtlsTransport and configuring the
> SrtpTransport.
>
> SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
>
> BUG=webrtc:7013
>
> Review-Url: https://codereview.webrtc.org/2997983002
> Cr-Commit-Position: refs/heads/master@{#19636}
> Committed: e683c6871fTBR=deadbeef@webrtc.org,pthatcher@google.com,zhihuang@webrtc.org
Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/3018513002
Cr-Commit-Position: refs/heads/master@{#19895}
Reason for revert:
Speculative revert since all Android bots on WebRTC FYI started to fail when this CL landed.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29
Original issue's description:
> If SRTP sessions exist, don't create new ones when applying answer.
>
> Instead, call the "Update" methods of SrtpSession, which will just call
> srtp_update, instead of wiping out the session state completely.
>
> This was causing decryption to stop working when subsequent
> offers/answers are applied. We don't know enough about SRTP to
> understand the root cause, and I wasn't able to write an integration
> test that reproduces the issue... But at least this fixes the bug that
> can be reproduced reliably using Hangouts.
>
> BUG=webrtc:8251
>
> Review-Url: https://codereview.webrtc.org/3019443002
> Cr-Commit-Position: refs/heads/master@{#19874}
> Committed: https://webrtc.googlesource.com/src/+/5ada7acf603e90e71990e9d4ff8f49389f24958cTBR=zhihuang@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8251
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/3017543002
Cr-Commit-Position: refs/heads/master@{#19882}
Instead, call the "Update" methods of SrtpSession, which will just call
srtp_update, instead of wiping out the session state completely.
This was causing decryption to stop working when subsequent
offers/answers are applied. We don't know enough about SRTP to
understand the root cause, and I wasn't able to write an integration
test that reproduces the issue... But at least this fixes the bug that
can be reproduced reliably using Hangouts.
BUG=webrtc:8251
Review-Url: https://codereview.webrtc.org/3019443002
Cr-Commit-Position: refs/heads/master@{#19874}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/pc/srtptransport.cc (Browse further)
