Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(476)

Side by Side Diff: webrtc/api/ortc/rtptransportinterface.h

Issue 2981513002: Wire up RTP keep-alive in ortc api. (Closed)
Patch Set: deps, again Created 3 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/api/ortc/ortcfactoryinterface.h ('k') | webrtc/call/BUILD.gn » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2017 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_ 11 #ifndef WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_
12 #define WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_ 12 #define WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_
13 13
14 #include <string> 14 #include <string>
15 15
16 #include "webrtc/api/ortc/packettransportinterface.h" 16 #include "webrtc/api/ortc/packettransportinterface.h"
17 #include "webrtc/api/rtcerror.h" 17 #include "webrtc/api/rtcerror.h"
18 #include "webrtc/common_types.h"
18 #include "webrtc/rtc_base/optional.h" 19 #include "webrtc/rtc_base/optional.h"
19 20
20 namespace webrtc { 21 namespace webrtc {
21 22
22 class RtpTransportAdapter; 23 class RtpTransportAdapter;
23 24
24 struct RtcpParameters { 25 struct RtcpParameters final {
25 // The SSRC to be used in the "SSRC of packet sender" field. If not set, one 26 // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
26 // will be chosen by the implementation. 27 // will be chosen by the implementation.
27 // TODO(deadbeef): Not implemented. 28 // TODO(deadbeef): Not implemented.
28 rtc::Optional<uint32_t> ssrc; 29 rtc::Optional<uint32_t> ssrc;
29 30
30 // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages). 31 // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
31 // 32 //
32 // If empty in the construction of the RtpTransport, one will be generated by 33 // If empty in the construction of the RtpTransport, one will be generated by
33 // the implementation, and returned in GetRtcpParameters. Multiple 34 // the implementation, and returned in GetRtcpParameters. Multiple
34 // RtpTransports created by the same OrtcFactory will use the same generated 35 // RtpTransports created by the same OrtcFactory will use the same generated
35 // CNAME. 36 // CNAME.
36 // 37 //
37 // If empty when passed into SetRtcpParameters, the CNAME simply won't be 38 // If empty when passed into SetParameters, the CNAME simply won't be
38 // modified. 39 // modified.
39 std::string cname; 40 std::string cname;
40 41
41 // Send reduced-size RTCP? 42 // Send reduced-size RTCP?
42 bool reduced_size = false; 43 bool reduced_size = false;
43 44
44 // Send RTCP multiplexed on the RTP transport? 45 // Send RTCP multiplexed on the RTP transport?
45 bool mux = true; 46 bool mux = true;
46 47
47 bool operator==(const RtcpParameters& o) const { 48 bool operator==(const RtcpParameters& o) const {
48 return ssrc == o.ssrc && cname == o.cname && 49 return ssrc == o.ssrc && cname == o.cname &&
49 reduced_size == o.reduced_size && mux == o.mux; 50 reduced_size == o.reduced_size && mux == o.mux;
50 } 51 }
51 bool operator!=(const RtcpParameters& o) const { return !(*this == o); } 52 bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
52 }; 53 };
53 54
55 struct RtpTransportParameters final {
56 RtcpParameters rtcp;
57
58 // Enabled periodic sending of keep-alive packets, that help prevent timeouts
59 // on the network level, such as NAT bindings. See RFC6263 section 4.6.
60 RtpKeepAliveConfig keepalive;
61
62 bool operator==(const RtpTransportParameters& o) const {
63 return rtcp == o.rtcp && keepalive == o.keepalive;
64 }
65 bool operator!=(const RtpTransportParameters& o) const {
66 return !(*this == o);
67 }
68 };
69
54 // Base class for different types of RTP transports that can be created by an 70 // Base class for different types of RTP transports that can be created by an
55 // OrtcFactory. Used by RtpSenders/RtpReceivers. 71 // OrtcFactory. Used by RtpSenders/RtpReceivers.
56 // 72 //
57 // This is not present in the standard ORTC API, but exists here for a few 73 // This is not present in the standard ORTC API, but exists here for a few
58 // reasons. Firstly, it allows different types of RTP transports to be used: 74 // reasons. Firstly, it allows different types of RTP transports to be used:
59 // DTLS-SRTP (which is required for the web), but also SDES-SRTP and 75 // DTLS-SRTP (which is required for the web), but also SDES-SRTP and
60 // unencrypted RTP. It also simplifies the handling of RTCP muxing, and 76 // unencrypted RTP. It also simplifies the handling of RTCP muxing, and
61 // provides a better API point for it. 77 // provides a better API point for it.
62 // 78 //
63 // Note that Edge's implementation of ORTC provides a similar API point, called 79 // Note that Edge's implementation of ORTC provides a similar API point, called
64 // RTCSrtpSdesTransport: 80 // RTCSrtpSdesTransport:
65 // https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx 81 // https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
66 class RtpTransportInterface { 82 class RtpTransportInterface {
67 public: 83 public:
68 virtual ~RtpTransportInterface() {} 84 virtual ~RtpTransportInterface() {}
69 85
70 // Returns packet transport that's used to send RTP packets. 86 // Returns packet transport that's used to send RTP packets.
71 virtual PacketTransportInterface* GetRtpPacketTransport() const = 0; 87 virtual PacketTransportInterface* GetRtpPacketTransport() const = 0;
72 88
73 // Returns separate packet transport that's used to send RTCP packets. If 89 // Returns separate packet transport that's used to send RTCP packets. If
74 // RTCP multiplexing is being used, returns null. 90 // RTCP multiplexing is being used, returns null.
75 virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0; 91 virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0;
76 92
77 // Set/get RTCP params. Can be used to enable RTCP muxing or reduced-size 93 // Set/get RTP/RTCP transport params. Can be used to enable RTCP muxing or
78 // RTCP if initially not enabled. 94 // reduced-size RTCP if initially not enabled.
79 // 95 //
80 // Changing |mux| from "true" to "false" is not allowed, and changing the 96 // Changing |mux| from "true" to "false" is not allowed, and changing the
81 // CNAME is currently unsupported. 97 // CNAME is currently unsupported.
82 virtual RTCError SetRtcpParameters(const RtcpParameters& parameters) = 0; 98 // RTP keep-alive settings need to be set before before an RtpSender has
99 // started sending, altering the payload type or timeout interval after this
100 // point is not supported. The parameters must also match across all RTP
101 // transports for a given RTP transport controller.
102 virtual RTCError SetParameters(const RtpTransportParameters& parameters) = 0;
83 // Returns last set or constructed-with parameters. If |cname| was empty in 103 // Returns last set or constructed-with parameters. If |cname| was empty in
84 // construction, the generated CNAME will be present in the returned 104 // construction, the generated CNAME will be present in the returned
85 // parameters (see above). 105 // parameters (see above).
86 virtual RtcpParameters GetRtcpParameters() const = 0; 106 virtual RtpTransportParameters GetParameters() const = 0;
87 107
88 protected: 108 protected:
89 // Only for internal use. Returns a pointer to an internal interface, for use 109 // Only for internal use. Returns a pointer to an internal interface, for use
90 // by the implementation. 110 // by the implementation.
91 virtual RtpTransportAdapter* GetInternal() = 0; 111 virtual RtpTransportAdapter* GetInternal() = 0;
92 112
93 // Classes that can use this internal interface. 113 // Classes that can use this internal interface.
94 friend class OrtcFactory; 114 friend class OrtcFactory;
95 friend class OrtcRtpSenderAdapter; 115 friend class OrtcRtpSenderAdapter;
96 friend class OrtcRtpReceiverAdapter; 116 friend class OrtcRtpReceiverAdapter;
97 friend class RtpTransportControllerAdapter; 117 friend class RtpTransportControllerAdapter;
98 friend class RtpTransportAdapter; 118 friend class RtpTransportAdapter;
99 }; 119 };
100 120
101 } // namespace webrtc 121 } // namespace webrtc
102 122
103 #endif // WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_ 123 #endif // WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_
OLDNEW
« no previous file with comments | « webrtc/api/ortc/ortcfactoryinterface.h ('k') | webrtc/call/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698