eb23e17798
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: e683c6871f
TBR=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}
63 lines
2.1 KiB
C++
63 lines
2.1 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/srtptransport.h"
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#include <string>
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#include "media/base/rtputils.h"
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#include "pc/rtptransport.h"
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#include "pc/srtpsession.h"
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#include "rtc_base/asyncpacketsocket.h"
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#include "rtc_base/copyonwritebuffer.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
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const std::string& content_name)
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: content_name_(content_name),
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rtp_transport_(rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled)) {
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ConnectToRtpTransport();
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}
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SrtpTransport::SrtpTransport(std::unique_ptr<RtpTransportInternal> transport,
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const std::string& content_name)
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: content_name_(content_name), rtp_transport_(std::move(transport)) {
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ConnectToRtpTransport();
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}
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void SrtpTransport::ConnectToRtpTransport() {
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rtp_transport_->SignalPacketReceived.connect(
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this, &SrtpTransport::OnPacketReceived);
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rtp_transport_->SignalReadyToSend.connect(this,
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&SrtpTransport::OnReadyToSend);
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}
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bool SrtpTransport::SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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// TODO(zstein): Protect packet.
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return rtp_transport_->SendPacket(rtcp, packet, options, flags);
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}
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void SrtpTransport::OnPacketReceived(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) {
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// TODO(zstein): Unprotect packet.
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SignalPacketReceived(rtcp, packet, packet_time);
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}
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} // namespace webrtc
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