This changes all internal code to use the media_description() helper
for ContentInfo along with the as_audio, as_video, and as_data casting
methods introduced in a previous CL. Reduces the total number of
pointer static_casts in pc/ from 351 to 122.
Bug: webrtc:8620
Change-Id: I996f49b55f1501c758a9e5223e30539a9f8d4eac
Reviewed-on: https://webrtc-review.googlesource.com/35921
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21419}
Eventually we want BaseChannel to depend on the RtpTransportInternal
instead of DtlsTransportInternal and share RtpTransport when bundling.
This CL is the first step.
Add SetRtpTransport and Init_w(RtptransportInternal*) to BaseChannel.
These two methods would replace the existing SetTransports and Init_w
methods.
Add new CreateVoice/VideoChannel methods to the ChannelManager which
take RtpTransportInternal instead of Dtls/PacketTransportInternal.
|cotnent_name| is removed from the SrtpTransport to simplify to code
since it is only used for debugging.
InitNetwork_n is removed from BaseChannel in CL as well.
Bug: webrtc:7013
Change-Id: I35b1565958548bd4896854c49e61d3ee160b7634
Reviewed-on: https://webrtc-review.googlesource.com/27840
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21057}
When enabling SDES in BaseChannel, SrtpTransport::EnableExternalAuth
should be called if the external authenication is enabled which is
not covered by the current test set.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ibbf458516a521a488e8e3bb4a5a29fca70a627f5
Reviewed-on: https://webrtc-review.googlesource.com/27761
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20960}
Add a method to DtlsSrtpTransport to cache the RTP Absolute Send Time
extension id. The method would be called when using DTLS-SRTP with
external authentication.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ie9edb9382cbb4cf43eea5da3030991a0d20293a5
Reviewed-on: https://webrtc-review.googlesource.com/27260
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20947}
The DtlsSrtpTransport takes the reponsiblity of setting up DTLS-SRTP from
the BaseChannel.
The BaseChannel doesn't handle the signals from the P2P layer transport anymore.
The RtpTransport handles the signals from the PacketTransportInternal and the
DtlsSrtpTransport handles the DTLS-specific signals and determines when to extract
the keys and setting the parameters.
In channel_unittests.cc, call from DTLS to SDES is expected to fail since the
fallback from DTLS to SDES is not supported.
Bug: webrtc:7013
Change-Id: I0a54e017986f5a8ae9710e79643a4651bef3c38f
Reviewed-on: https://webrtc-review.googlesource.com/24702
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20941}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: If41c462dc3ddff664d0b70d249d760e2ca4c8ab3
Reviewed-on: https://webrtc-review.googlesource.com/23576
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20820}
In BaseChannel, |selected_candidate_pair| is removed.
In MediaContentDescription, |buffered_mode_latency_| and its
getter/setter are removed.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: I68c0a61136dcd078f587105f09c72098d7f8e620
Reviewed-on: https://webrtc-review.googlesource.com/23520
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20694}
The downstream application doesn't use CA_UPDATE and the related code are
removed to simplify the BaseChannel.
TBR=pthatcher@webrtc.org
Bug: webrtc:8521
Change-Id: I9adc1539db7feb7b5c3aafba7a2be7100f2c068a
Reviewed-on: https://webrtc-review.googlesource.com/22205
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20693}
OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.
This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520
Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20678}
|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}
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}
Always use the PF_SRTP_BYPASS when sending RTP/RTCP packets.
Change the condition in BaseChannel::GetSrtpOverhead.
Get the SRTP overhead when using either SDES or DTLS-SRTP.
Bug: None
Change-Id: I44aeff8b75e56b12acefd73299a95a3e38cd401b
Reviewed-on: https://webrtc-review.googlesource.com/6580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20152}
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}
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}
