1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
|
/*
* 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 API_ORTC_RTPTRANSPORTINTERFACE_H_
#define API_ORTC_RTPTRANSPORTINTERFACE_H_
#include <string>
#include "absl/types/optional.h"
#include "api/ortc/packettransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtp_headers.h"
#include "api/rtpparameters.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
class RtpTransportAdapter;
struct RtpTransportParameters final {
RtcpParameters rtcp;
// Enabled periodic sending of keep-alive packets, that help prevent timeouts
// on the network level, such as NAT bindings. See RFC6263 section 4.6.
RtpKeepAliveConfig keepalive;
bool operator==(const RtpTransportParameters& o) const {
return rtcp == o.rtcp && keepalive == o.keepalive;
}
bool operator!=(const RtpTransportParameters& o) const {
return !(*this == o);
}
};
// Base class for different types of RTP transports that can be created by an
// OrtcFactory. Used by RtpSenders/RtpReceivers.
//
// This is not present in the standard ORTC API, but exists here for a few
// reasons. Firstly, it allows different types of RTP transports to be used:
// DTLS-SRTP (which is required for the web), but also SDES-SRTP and
// unencrypted RTP. It also simplifies the handling of RTCP muxing, and
// provides a better API point for it.
//
// Note that Edge's implementation of ORTC provides a similar API point, called
// RTCSrtpSdesTransport:
// https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
class RtpTransportInterface {
public:
virtual ~RtpTransportInterface() {}
// Returns packet transport that's used to send RTP packets.
virtual PacketTransportInterface* GetRtpPacketTransport() const = 0;
// Returns separate packet transport that's used to send RTCP packets. If
// RTCP multiplexing is being used, returns null.
virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0;
// Set/get RTP/RTCP transport params. Can be used to enable RTCP muxing or
// reduced-size RTCP if initially not enabled.
//
// Changing |mux| from "true" to "false" is not allowed, and changing the
// CNAME is currently unsupported.
// RTP keep-alive settings need to be set before before an RtpSender has
// started sending, altering the payload type or timeout interval after this
// point is not supported. The parameters must also match across all RTP
// transports for a given RTP transport controller.
virtual RTCError SetParameters(const RtpTransportParameters& parameters) = 0;
// Returns last set or constructed-with parameters. If |cname| was empty in
// construction, the generated CNAME will be present in the returned
// parameters (see above).
virtual RtpTransportParameters GetParameters() const = 0;
protected:
// Only for internal use. Returns a pointer to an internal interface, for use
// by the implementation.
virtual RtpTransportAdapter* GetInternal() = 0;
// Classes that can use this internal interface.
friend class OrtcFactory;
friend class OrtcRtpSenderAdapter;
friend class OrtcRtpReceiverAdapter;
friend class RtpTransportControllerAdapter;
friend class RtpTransportAdapter;
};
} // namespace webrtc
#endif // API_ORTC_RTPTRANSPORTINTERFACE_H_
|
