106 lines
4.0 KiB
C
106 lines
4.0 KiB
C
|
/*
|
||
|
|
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
|
||
|
|
*
|
||
|
|
* Use of this source code is governed by a BSD-style license
|
||
|
|
* that can be found in the LICENSE file in the root of the source
|
||
|
|
* tree. An additional intellectual property rights grant can be found
|
||
|
|
* in the file PATENTS. All contributing project authors may
|
||
|
|
* be found in the AUTHORS file in the root of the source tree.
|
||
|
|
*/
|
||
|
|
|
||
|
|
#ifndef PC_RTP_TRANSPORT_INTERNAL_H_
|
||
|
|
#define PC_RTP_TRANSPORT_INTERNAL_H_
|
||
|
|
|
||
|
|
#include <string>
|
||
|
|
|
||
|
|
#include "call/rtp_demuxer.h"
|
||
|
|
#include "p2p/base/ice_transport_internal.h"
|
||
|
|
#include "pc/session_description.h"
|
||
|
|
#include "rtc_base/network_route.h"
|
||
|
|
#include "rtc_base/ssl_stream_adapter.h"
|
||
|
|
#include "rtc_base/third_party/sigslot/sigslot.h"
|
||
|
|
|
||
|
|
namespace rtc {
|
||
|
|
class CopyOnWriteBuffer;
|
||
|
|
struct PacketOptions;
|
||
|
|
} // namespace rtc
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
|
||
|
|
// This represents the internal interface beneath SrtpTransportInterface;
|
||
|
|
// it is not accessible to API consumers but is accessible to internal classes
|
||
|
|
// in order to send and receive RTP and RTCP packets belonging to a single RTP
|
||
|
|
// session. Additional convenience and configuration methods are also provided.
|
||
|
|
class RtpTransportInternal : public sigslot::has_slots<> {
|
||
|
|
public:
|
||
|
|
virtual ~RtpTransportInternal() = default;
|
||
|
|
|
||
|
|
virtual void SetRtcpMuxEnabled(bool enable) = 0;
|
||
|
|
|
||
|
|
virtual const std::string& transport_name() const = 0;
|
||
|
|
|
||
|
|
// Sets socket options on the underlying RTP or RTCP transports.
|
||
|
|
virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0;
|
||
|
|
virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0;
|
||
|
|
|
||
|
|
virtual bool rtcp_mux_enabled() const = 0;
|
||
|
|
|
||
|
|
virtual bool IsReadyToSend() const = 0;
|
||
|
|
|
||
|
|
// Called whenever a transport's ready-to-send state changes. The argument
|
||
|
|
// is true if all used transports are ready to send. This is more specific
|
||
|
|
// than just "writable"; it means the last send didn't return ENOTCONN.
|
||
|
|
sigslot::signal1<bool> SignalReadyToSend;
|
||
|
|
|
||
|
|
// Called whenever an RTCP packet is received. There is no equivalent signal
|
||
|
|
// for RTP packets because they would be forwarded to the BaseChannel through
|
||
|
|
// the RtpDemuxer callback.
|
||
|
|
sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
|
||
|
|
|
||
|
|
// Called whenever the network route of the P2P layer transport changes.
|
||
|
|
// The argument is an optional network route.
|
||
|
|
sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
|
||
|
|
|
||
|
|
// Called whenever a transport's writable state might change. The argument is
|
||
|
|
// true if the transport is writable, otherwise it is false.
|
||
|
|
sigslot::signal1<bool> SignalWritableState;
|
||
|
|
|
||
|
|
sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
|
||
|
|
|
||
|
|
virtual bool IsWritable(bool rtcp) const = 0;
|
||
|
|
|
||
|
|
// TODO(zhihuang): Pass the |packet| by copy so that the original data
|
||
|
|
// wouldn't be modified.
|
||
|
|
virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
|
||
|
|
const rtc::PacketOptions& options,
|
||
|
|
int flags) = 0;
|
||
|
|
|
||
|
|
virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
|
||
|
|
const rtc::PacketOptions& options,
|
||
|
|
int flags) = 0;
|
||
|
|
|
||
|
|
// This method updates the RTP header extension map so that the RTP transport
|
||
|
|
// can parse the received packets and identify the MID. This is called by the
|
||
|
|
// BaseChannel when setting the content description.
|
||
|
|
//
|
||
|
|
// TODO(zhihuang): Merging and replacing following methods handling header
|
||
|
|
// extensions with SetParameters:
|
||
|
|
// UpdateRtpHeaderExtensionMap,
|
||
|
|
// UpdateSendEncryptedHeaderExtensionIds,
|
||
|
|
// UpdateRecvEncryptedHeaderExtensionIds,
|
||
|
|
// CacheRtpAbsSendTimeHeaderExtension,
|
||
|
|
virtual void UpdateRtpHeaderExtensionMap(
|
||
|
|
const cricket::RtpHeaderExtensions& header_extensions) = 0;
|
||
|
|
|
||
|
|
virtual bool IsSrtpActive() const = 0;
|
||
|
|
|
||
|
|
virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
|
||
|
|
RtpPacketSinkInterface* sink) = 0;
|
||
|
|
|
||
|
|
virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace webrtc
|
||
|
|
|
||
|
|
#endif // PC_RTP_TRANSPORT_INTERNAL_H_
|