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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_PC_SRTPTRANSPORT_H_ | 11 #ifndef WEBRTC_PC_SRTPTRANSPORT_H_ |
12 #define WEBRTC_PC_SRTPTRANSPORT_H_ | 12 #define WEBRTC_PC_SRTPTRANSPORT_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <utility> | 16 #include <utility> |
17 | 17 |
18 #include "webrtc/pc/rtptransportinternal.h" | 18 #include "webrtc/pc/rtptransportinternal.h" |
19 #include "webrtc/pc/srtpfilter.h" | 19 #include "webrtc/pc/srtpfilter.h" |
20 #include "webrtc/pc/srtpsession.h" | |
20 #include "webrtc/rtc_base/checks.h" | 21 #include "webrtc/rtc_base/checks.h" |
21 | 22 |
22 namespace webrtc { | 23 namespace webrtc { |
23 | 24 |
24 // This class will eventually be a wrapper around RtpTransportInternal | 25 // This class will eventually be a wrapper around RtpTransportInternal |
25 // that protects and unprotects sent and received RTP packets. This | 26 // that protects and unprotects sent and received RTP packets. This |
26 // functionality is currently implemented by SrtpFilter and BaseChannel, but | 27 // functionality is currently implemented by SrtpFilter and BaseChannel, but |
27 // will be moved here in the future. | 28 // will be moved here in the future. |
28 class SrtpTransport : public RtpTransportInternal { | 29 class SrtpTransport : public RtpTransportInternal { |
29 public: | 30 public: |
(...skipping 28 matching lines...) Expand all Loading... | |
58 } | 59 } |
59 | 60 |
60 PacketTransportInterface* GetRtcpPacketTransport() const override { | 61 PacketTransportInterface* GetRtcpPacketTransport() const override { |
61 return rtp_transport_->GetRtcpPacketTransport(); | 62 return rtp_transport_->GetRtcpPacketTransport(); |
62 } | 63 } |
63 | 64 |
64 bool IsWritable(bool rtcp) const override { | 65 bool IsWritable(bool rtcp) const override { |
65 return rtp_transport_->IsWritable(rtcp); | 66 return rtp_transport_->IsWritable(rtcp); |
66 } | 67 } |
67 | 68 |
69 // The transport becomes active if the send_session_ and recv_session_ are | |
70 // created. | |
71 bool IsActive() const; | |
72 | |
68 bool SendPacket(bool rtcp, | 73 bool SendPacket(bool rtcp, |
69 rtc::CopyOnWriteBuffer* packet, | 74 rtc::CopyOnWriteBuffer* packet, |
70 const rtc::PacketOptions& options, | 75 const rtc::PacketOptions& options, |
71 int flags) override; | 76 int flags) override; |
72 | 77 |
73 bool HandlesPayloadType(int payload_type) const override { | 78 bool HandlesPayloadType(int payload_type) const override { |
74 return rtp_transport_->HandlesPayloadType(payload_type); | 79 return rtp_transport_->HandlesPayloadType(payload_type); |
75 } | 80 } |
76 | 81 |
77 void AddHandledPayloadType(int payload_type) override { | 82 void AddHandledPayloadType(int payload_type) override { |
78 rtp_transport_->AddHandledPayloadType(payload_type); | 83 rtp_transport_->AddHandledPayloadType(payload_type); |
79 } | 84 } |
80 | 85 |
81 RtcpParameters GetRtcpParameters() const override { | 86 RtcpParameters GetRtcpParameters() const override { |
82 return rtp_transport_->GetRtcpParameters(); | 87 return rtp_transport_->GetRtcpParameters(); |
83 } | 88 } |
84 | 89 |
85 RTCError SetRtcpParameters(const RtcpParameters& parameters) override { | 90 RTCError SetRtcpParameters(const RtcpParameters& parameters) override { |
86 return rtp_transport_->SetRtcpParameters(parameters); | 91 return rtp_transport_->SetRtcpParameters(parameters); |
87 } | 92 } |
88 | 93 |
89 // TODO(zstein): Remove this when we remove RtpTransportAdapter. | 94 // TODO(zstein): Remove this when we remove RtpTransportAdapter. |
90 RtpTransportAdapter* GetInternal() override { return nullptr; } | 95 RtpTransportAdapter* GetInternal() override { return nullptr; } |
91 | 96 |
97 // Create new send/recv sessions and set the negotiated crypto keys for RTP | |
98 // packet encryption. The keys can either come from SDES negotiation or DTLS | |
99 // handshake. | |
100 bool SetRtpParams(int send_cs, | |
101 const uint8_t* send_key, | |
102 int send_key_len, | |
103 int recv_cs, | |
104 const uint8_t* recv_key, | |
105 int recv_key_len); | |
106 | |
107 // Create new send/recv sessions and set the negotiated crypto keys for RTCP | |
108 // packet encryption. The keys can either come from SDES negotiation or DTLS | |
109 // handshake. | |
110 bool SetRtcpParams(int send_cs, | |
111 const uint8_t* send_key, | |
112 int send_key_len, | |
113 int recv_cs, | |
114 const uint8_t* recv_key, | |
115 int recv_key_len); | |
116 | |
117 // When the send/recv sessions have been created, just updated the crypto | |
118 // keys. | |
pthatcher
2017/08/28 21:42:56
Why do we have this split between SetRtpParams and
Zhi Huang
2017/08/29 18:40:35
SetRtpParams calls SrtpSession::SetSend/Recv;
Upd
| |
119 bool UpdateRtpParams(int send_cs, | |
120 const uint8_t* send_key, | |
121 int send_key_len, | |
122 int recv_cs, | |
123 const uint8_t* recv_key, | |
124 int recv_key_len); | |
125 | |
126 void ResetParams(); | |
127 | |
128 // Set the header extension ids that should be encrypted for the given source. | |
129 void SetEncryptedHeaderExtensionIds(cricket::ContentSource source, | |
130 const std::vector<int>& extension_ids); | |
131 | |
132 // If external auth is enabled, SRTP will write a dummy auth tag that then | |
133 // later must get replaced before the packet is sent out. Only supported for | |
134 // non-GCM cipher suites and can be checked through "IsExternalAuthActive" | |
135 // if it is actually used. This method is only valid before the RTP params | |
136 // have been set. | |
137 void EnableExternalAuth(); | |
138 bool IsExternalAuthEnabled() const; | |
139 | |
140 // A SrtpTransport supports external creation of the auth tag if a non-GCM | |
141 // cipher is used. This method is only valid after the RTP params have | |
142 // been set. | |
143 bool IsExternalAuthActive() const; | |
144 | |
145 // Returns srtp overhead for rtp packets. | |
146 bool GetSrtpOverhead(int* srtp_overhead) const; | |
147 | |
148 // Returns rtp auth params from srtp context. | |
149 bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len); | |
150 | |
151 // Helper method to get RTP Absoulute SendTime extension header id if | |
152 // present in remote supported extensions list. | |
153 void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) { | |
154 rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id; | |
155 } | |
156 | |
92 private: | 157 private: |
158 void CreateSrtpSessions(); | |
159 | |
93 void ConnectToRtpTransport(); | 160 void ConnectToRtpTransport(); |
94 | 161 |
95 void OnPacketReceived(bool rtcp, | 162 void OnPacketReceived(bool rtcp, |
96 rtc::CopyOnWriteBuffer* packet, | 163 rtc::CopyOnWriteBuffer* packet, |
97 const rtc::PacketTime& packet_time); | 164 const rtc::PacketTime& packet_time); |
98 | 165 |
99 void OnReadyToSend(bool ready) { SignalReadyToSend(ready); } | 166 void OnReadyToSend(bool ready) { SignalReadyToSend(ready); } |
100 | 167 |
168 bool ProtectRtp(void* data, int in_len, int max_len, int* out_len); | |
169 | |
170 // Overloaded version, outputs packet index. | |
171 bool ProtectRtp(void* data, | |
172 int in_len, | |
173 int max_len, | |
174 int* out_len, | |
175 int64_t* index); | |
176 bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len); | |
177 | |
178 // Decrypts/verifies an invidiual RTP/RTCP packet. | |
179 // If an HMAC is used, this will decrease the packet size. | |
180 bool UnprotectRtp(void* data, int in_len, int* out_len); | |
181 | |
182 bool UnprotectRtcp(void* data, int in_len, int* out_len); | |
183 | |
101 const std::string content_name_; | 184 const std::string content_name_; |
185 std::unique_ptr<RtpTransportInternal> rtp_transport_; | |
102 | 186 |
103 std::unique_ptr<RtpTransportInternal> rtp_transport_; | 187 std::unique_ptr<cricket::SrtpSession> send_session_; |
188 std::unique_ptr<cricket::SrtpSession> recv_session_; | |
189 std::unique_ptr<cricket::SrtpSession> send_rtcp_session_; | |
190 std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_; | |
191 | |
192 std::vector<int> send_encrypted_header_extension_ids_; | |
193 std::vector<int> recv_encrypted_header_extension_ids_; | |
194 bool external_auth_enabled_ = false; | |
195 | |
196 int rtp_abs_sendtime_extn_id_ = -1; | |
104 }; | 197 }; |
105 | 198 |
106 } // namespace webrtc | 199 } // namespace webrtc |
107 | 200 |
108 #endif // WEBRTC_PC_SRTPTRANSPORT_H_ | 201 #endif // WEBRTC_PC_SRTPTRANSPORT_H_ |
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