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1 /* | |
2 * libjingle | |
3 * Copyright 2009 Google Inc. | |
4 * | |
5 * Redistribution and use in source and binary forms, with or without | |
6 * modification, are permitted provided that the following conditions are met: | |
7 * | |
8 * 1. Redistributions of source code must retain the above copyright notice, | |
9 * this list of conditions and the following disclaimer. | |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
11 * this list of conditions and the following disclaimer in the documentation | |
12 * and/or other materials provided with the distribution. | |
13 * 3. The name of the author may not be used to endorse or promote products | |
14 * derived from this software without specific prior written permission. | |
15 * | |
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
26 */ | |
27 | |
28 #ifndef TALK_SESSION_MEDIA_SRTPFILTER_H_ | |
29 #define TALK_SESSION_MEDIA_SRTPFILTER_H_ | |
30 | |
31 #include <list> | |
32 #include <map> | |
33 #include <string> | |
34 #include <vector> | |
35 | |
36 #include "webrtc/base/basictypes.h" | |
37 #include "webrtc/base/criticalsection.h" | |
38 #include "webrtc/base/scoped_ptr.h" | |
39 #include "webrtc/base/sigslotrepeater.h" | |
40 #include "webrtc/base/sslstreamadapter.h" | |
41 #include "webrtc/media/base/cryptoparams.h" | |
42 #include "webrtc/p2p/base/sessiondescription.h" | |
43 | |
44 // Forward declaration to avoid pulling in libsrtp headers here | |
45 struct srtp_event_data_t; | |
46 struct srtp_ctx_t; | |
47 struct srtp_policy_t; | |
48 | |
49 namespace cricket { | |
50 | |
51 // Key is 128 bits and salt is 112 bits == 30 bytes. B64 bloat => 40 bytes. | |
52 extern const int SRTP_MASTER_KEY_BASE64_LEN; | |
53 | |
54 // Needed for DTLS-SRTP | |
55 extern const int SRTP_MASTER_KEY_KEY_LEN; | |
56 extern const int SRTP_MASTER_KEY_SALT_LEN; | |
57 | |
58 class SrtpSession; | |
59 class SrtpStat; | |
60 | |
61 void EnableSrtpDebugging(); | |
62 void ShutdownSrtp(); | |
63 | |
64 // Class to transform SRTP to/from RTP. | |
65 // Initialize by calling SetSend with the local security params, then call | |
66 // SetRecv once the remote security params are received. At that point | |
67 // Protect/UnprotectRt(c)p can be called to encrypt/decrypt data. | |
68 // TODO: Figure out concurrency policy for SrtpFilter. | |
69 class SrtpFilter { | |
70 public: | |
71 enum Mode { | |
72 PROTECT, | |
73 UNPROTECT | |
74 }; | |
75 enum Error { | |
76 ERROR_NONE, | |
77 ERROR_FAIL, | |
78 ERROR_AUTH, | |
79 ERROR_REPLAY, | |
80 }; | |
81 | |
82 SrtpFilter(); | |
83 ~SrtpFilter(); | |
84 | |
85 // Whether the filter is active (i.e. crypto has been properly negotiated). | |
86 bool IsActive() const; | |
87 | |
88 // Indicates which crypto algorithms and keys were contained in the offer. | |
89 // offer_params should contain a list of available parameters to use, or none, | |
90 // if crypto is not desired. This must be called before SetAnswer. | |
91 bool SetOffer(const std::vector<CryptoParams>& offer_params, | |
92 ContentSource source); | |
93 // Same as SetAnwer. But multiple calls are allowed to SetProvisionalAnswer | |
94 // after a call to SetOffer. | |
95 bool SetProvisionalAnswer(const std::vector<CryptoParams>& answer_params, | |
96 ContentSource source); | |
97 // Indicates which crypto algorithms and keys were contained in the answer. | |
98 // answer_params should contain the negotiated parameters, which may be none, | |
99 // if crypto was not desired or could not be negotiated (and not required). | |
100 // This must be called after SetOffer. If crypto negotiation completes | |
101 // successfully, this will advance the filter to the active state. | |
102 bool SetAnswer(const std::vector<CryptoParams>& answer_params, | |
103 ContentSource source); | |
104 | |
105 // Just set up both sets of keys directly. | |
106 // Used with DTLS-SRTP. | |
107 bool SetRtpParams(int send_cs, | |
108 const uint8_t* send_key, | |
109 int send_key_len, | |
110 int recv_cs, | |
111 const uint8_t* recv_key, | |
112 int recv_key_len); | |
113 bool SetRtcpParams(int send_cs, | |
114 const uint8_t* send_key, | |
115 int send_key_len, | |
116 int recv_cs, | |
117 const uint8_t* recv_key, | |
118 int recv_key_len); | |
119 | |
120 // Encrypts/signs an individual RTP/RTCP packet, in-place. | |
121 // If an HMAC is used, this will increase the packet size. | |
122 bool ProtectRtp(void* data, int in_len, int max_len, int* out_len); | |
123 // Overloaded version, outputs packet index. | |
124 bool ProtectRtp(void* data, | |
125 int in_len, | |
126 int max_len, | |
127 int* out_len, | |
128 int64_t* index); | |
129 bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len); | |
130 // Decrypts/verifies an invidiual RTP/RTCP packet. | |
131 // If an HMAC is used, this will decrease the packet size. | |
132 bool UnprotectRtp(void* data, int in_len, int* out_len); | |
133 bool UnprotectRtcp(void* data, int in_len, int* out_len); | |
134 | |
135 // Returns rtp auth params from srtp context. | |
136 bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len); | |
137 | |
138 // Update the silent threshold (in ms) for signaling errors. | |
139 void set_signal_silent_time(uint32_t signal_silent_time_in_ms); | |
140 | |
141 bool ResetParams(); | |
142 | |
143 sigslot::repeater3<uint32_t, Mode, Error> SignalSrtpError; | |
144 | |
145 protected: | |
146 bool ExpectOffer(ContentSource source); | |
147 bool StoreParams(const std::vector<CryptoParams>& params, | |
148 ContentSource source); | |
149 bool ExpectAnswer(ContentSource source); | |
150 bool DoSetAnswer(const std::vector<CryptoParams>& answer_params, | |
151 ContentSource source, | |
152 bool final); | |
153 void CreateSrtpSessions(); | |
154 bool NegotiateParams(const std::vector<CryptoParams>& answer_params, | |
155 CryptoParams* selected_params); | |
156 bool ApplyParams(const CryptoParams& send_params, | |
157 const CryptoParams& recv_params); | |
158 static bool ParseKeyParams(const std::string& params, uint8_t* key, int len); | |
159 | |
160 private: | |
161 enum State { | |
162 ST_INIT, // SRTP filter unused. | |
163 ST_SENTOFFER, // Offer with SRTP parameters sent. | |
164 ST_RECEIVEDOFFER, // Offer with SRTP parameters received. | |
165 ST_SENTPRANSWER_NO_CRYPTO, // Sent provisional answer without crypto. | |
166 // Received provisional answer without crypto. | |
167 ST_RECEIVEDPRANSWER_NO_CRYPTO, | |
168 ST_ACTIVE, // Offer and answer set. | |
169 // SRTP filter is active but new parameters are offered. | |
170 // When the answer is set, the state transitions to ST_ACTIVE or ST_INIT. | |
171 ST_SENTUPDATEDOFFER, | |
172 // SRTP filter is active but new parameters are received. | |
173 // When the answer is set, the state transitions back to ST_ACTIVE. | |
174 ST_RECEIVEDUPDATEDOFFER, | |
175 // SRTP filter is active but the sent answer is only provisional. | |
176 // When the final answer is set, the state transitions to ST_ACTIVE or | |
177 // ST_INIT. | |
178 ST_SENTPRANSWER, | |
179 // SRTP filter is active but the received answer is only provisional. | |
180 // When the final answer is set, the state transitions to ST_ACTIVE or | |
181 // ST_INIT. | |
182 ST_RECEIVEDPRANSWER | |
183 }; | |
184 State state_; | |
185 uint32_t signal_silent_time_in_ms_; | |
186 std::vector<CryptoParams> offer_params_; | |
187 rtc::scoped_ptr<SrtpSession> send_session_; | |
188 rtc::scoped_ptr<SrtpSession> recv_session_; | |
189 rtc::scoped_ptr<SrtpSession> send_rtcp_session_; | |
190 rtc::scoped_ptr<SrtpSession> recv_rtcp_session_; | |
191 CryptoParams applied_send_params_; | |
192 CryptoParams applied_recv_params_; | |
193 }; | |
194 | |
195 // Class that wraps a libSRTP session. | |
196 class SrtpSession { | |
197 public: | |
198 SrtpSession(); | |
199 ~SrtpSession(); | |
200 | |
201 // Configures the session for sending data using the specified | |
202 // cipher-suite and key. Receiving must be done by a separate session. | |
203 bool SetSend(int cs, const uint8_t* key, int len); | |
204 // Configures the session for receiving data using the specified | |
205 // cipher-suite and key. Sending must be done by a separate session. | |
206 bool SetRecv(int cs, const uint8_t* key, int len); | |
207 | |
208 // Encrypts/signs an individual RTP/RTCP packet, in-place. | |
209 // If an HMAC is used, this will increase the packet size. | |
210 bool ProtectRtp(void* data, int in_len, int max_len, int* out_len); | |
211 // Overloaded version, outputs packet index. | |
212 bool ProtectRtp(void* data, | |
213 int in_len, | |
214 int max_len, | |
215 int* out_len, | |
216 int64_t* index); | |
217 bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len); | |
218 // Decrypts/verifies an invidiual RTP/RTCP packet. | |
219 // If an HMAC is used, this will decrease the packet size. | |
220 bool UnprotectRtp(void* data, int in_len, int* out_len); | |
221 bool UnprotectRtcp(void* data, int in_len, int* out_len); | |
222 | |
223 // Helper method to get authentication params. | |
224 bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len); | |
225 | |
226 // Update the silent threshold (in ms) for signaling errors. | |
227 void set_signal_silent_time(uint32_t signal_silent_time_in_ms); | |
228 | |
229 // Calls srtp_shutdown if it's initialized. | |
230 static void Terminate(); | |
231 | |
232 sigslot::repeater3<uint32_t, SrtpFilter::Mode, SrtpFilter::Error> | |
233 SignalSrtpError; | |
234 | |
235 private: | |
236 bool SetKey(int type, int cs, const uint8_t* key, int len); | |
237 // Returns send stream current packet index from srtp db. | |
238 bool GetSendStreamPacketIndex(void* data, int in_len, int64_t* index); | |
239 | |
240 static bool Init(); | |
241 void HandleEvent(const srtp_event_data_t* ev); | |
242 static void HandleEventThunk(srtp_event_data_t* ev); | |
243 | |
244 static std::list<SrtpSession*>* sessions(); | |
245 | |
246 srtp_ctx_t* session_; | |
247 int rtp_auth_tag_len_; | |
248 int rtcp_auth_tag_len_; | |
249 rtc::scoped_ptr<SrtpStat> srtp_stat_; | |
250 static bool inited_; | |
251 static rtc::GlobalLockPod lock_; | |
252 int last_send_seq_num_; | |
253 RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession); | |
254 }; | |
255 | |
256 // Class that collects failures of SRTP. | |
257 class SrtpStat { | |
258 public: | |
259 SrtpStat(); | |
260 | |
261 // Report RTP protection results to the handler. | |
262 void AddProtectRtpResult(uint32_t ssrc, int result); | |
263 // Report RTP unprotection results to the handler. | |
264 void AddUnprotectRtpResult(uint32_t ssrc, int result); | |
265 // Report RTCP protection results to the handler. | |
266 void AddProtectRtcpResult(int result); | |
267 // Report RTCP unprotection results to the handler. | |
268 void AddUnprotectRtcpResult(int result); | |
269 | |
270 // Get silent time (in ms) for SRTP statistics handler. | |
271 uint32_t signal_silent_time() const { return signal_silent_time_; } | |
272 // Set silent time (in ms) for SRTP statistics handler. | |
273 void set_signal_silent_time(uint32_t signal_silent_time) { | |
274 signal_silent_time_ = signal_silent_time; | |
275 } | |
276 | |
277 // Sigslot for reporting errors. | |
278 sigslot::signal3<uint32_t, SrtpFilter::Mode, SrtpFilter::Error> | |
279 SignalSrtpError; | |
280 | |
281 private: | |
282 // For each different ssrc and error, we collect statistics separately. | |
283 struct FailureKey { | |
284 FailureKey() | |
285 : ssrc(0), | |
286 mode(SrtpFilter::PROTECT), | |
287 error(SrtpFilter::ERROR_NONE) { | |
288 } | |
289 FailureKey(uint32_t in_ssrc, | |
290 SrtpFilter::Mode in_mode, | |
291 SrtpFilter::Error in_error) | |
292 : ssrc(in_ssrc), mode(in_mode), error(in_error) {} | |
293 bool operator <(const FailureKey& key) const { | |
294 return | |
295 (ssrc < key.ssrc) || | |
296 (ssrc == key.ssrc && mode < key.mode) || | |
297 (ssrc == key.ssrc && mode == key.mode && error < key.error); | |
298 } | |
299 uint32_t ssrc; | |
300 SrtpFilter::Mode mode; | |
301 SrtpFilter::Error error; | |
302 }; | |
303 // For tracing conditions for signaling, currently we only use | |
304 // last_signal_time. Wrap this as a struct so that later on, if we need any | |
305 // other improvements, it will be easier. | |
306 struct FailureStat { | |
307 FailureStat() | |
308 : last_signal_time(0) { | |
309 } | |
310 explicit FailureStat(uint32_t in_last_signal_time) | |
311 : last_signal_time(in_last_signal_time) {} | |
312 void Reset() { | |
313 last_signal_time = 0; | |
314 } | |
315 uint32_t last_signal_time; | |
316 }; | |
317 | |
318 // Inspect SRTP result and signal error if needed. | |
319 void HandleSrtpResult(const FailureKey& key); | |
320 | |
321 std::map<FailureKey, FailureStat> failures_; | |
322 // Threshold in ms to silent the signaling errors. | |
323 uint32_t signal_silent_time_; | |
324 | |
325 RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat); | |
326 }; | |
327 | |
328 } // namespace cricket | |
329 | |
330 #endif // TALK_SESSION_MEDIA_SRTPFILTER_H_ | |
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