Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(602)

Side by Side Diff: talk/session/media/srtpfilter.h

Issue 1691463002: Move talk/session/media -> webrtc/pc (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Last rebase Created 4 years, 10 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 | « talk/session/media/rtcpmuxfilter_unittest.cc ('k') | talk/session/media/srtpfilter.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
(Empty)
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_
OLDNEW
« no previous file with comments | « talk/session/media/rtcpmuxfilter_unittest.cc ('k') | talk/session/media/srtpfilter.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698