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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2009 Google Inc. | 3 * Copyright 2009 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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32 #include <map> | 32 #include <map> |
33 #include <string> | 33 #include <string> |
34 #include <vector> | 34 #include <vector> |
35 | 35 |
36 #include "talk/media/base/cryptoparams.h" | 36 #include "talk/media/base/cryptoparams.h" |
37 #include "webrtc/p2p/base/sessiondescription.h" | 37 #include "webrtc/p2p/base/sessiondescription.h" |
38 #include "webrtc/base/basictypes.h" | 38 #include "webrtc/base/basictypes.h" |
39 #include "webrtc/base/criticalsection.h" | 39 #include "webrtc/base/criticalsection.h" |
40 #include "webrtc/base/scoped_ptr.h" | 40 #include "webrtc/base/scoped_ptr.h" |
41 #include "webrtc/base/sigslotrepeater.h" | 41 #include "webrtc/base/sigslotrepeater.h" |
42 #include "webrtc/base/sslstreamadapter.h" | |
43 | 42 |
44 // Forward declaration to avoid pulling in libsrtp headers here | 43 // Forward declaration to avoid pulling in libsrtp headers here |
45 struct srtp_event_data_t; | 44 struct srtp_event_data_t; |
46 struct srtp_ctx_t; | 45 struct srtp_ctx_t; |
47 struct srtp_policy_t; | 46 struct srtp_policy_t; |
48 | 47 |
49 namespace cricket { | 48 namespace cricket { |
50 | 49 |
| 50 // Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except |
| 51 // in applications (voice) where the additional bandwidth may be significant. |
| 52 // A 80-bit HMAC is always used for SRTCP. |
| 53 // 128-bit AES with 80-bit SHA-1 HMAC. |
| 54 extern const char CS_AES_CM_128_HMAC_SHA1_80[]; |
| 55 // 128-bit AES with 32-bit SHA-1 HMAC. |
| 56 extern const char CS_AES_CM_128_HMAC_SHA1_32[]; |
51 // Key is 128 bits and salt is 112 bits == 30 bytes. B64 bloat => 40 bytes. | 57 // Key is 128 bits and salt is 112 bits == 30 bytes. B64 bloat => 40 bytes. |
52 extern const int SRTP_MASTER_KEY_BASE64_LEN; | 58 extern const int SRTP_MASTER_KEY_BASE64_LEN; |
53 | 59 |
54 // Needed for DTLS-SRTP | 60 // Needed for DTLS-SRTP |
55 extern const int SRTP_MASTER_KEY_KEY_LEN; | 61 extern const int SRTP_MASTER_KEY_KEY_LEN; |
56 extern const int SRTP_MASTER_KEY_SALT_LEN; | 62 extern const int SRTP_MASTER_KEY_SALT_LEN; |
57 | 63 |
58 class SrtpSession; | 64 class SrtpSession; |
59 class SrtpStat; | 65 class SrtpStat; |
60 | 66 |
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313 std::map<FailureKey, FailureStat> failures_; | 319 std::map<FailureKey, FailureStat> failures_; |
314 // Threshold in ms to silent the signaling errors. | 320 // Threshold in ms to silent the signaling errors. |
315 uint32 signal_silent_time_; | 321 uint32 signal_silent_time_; |
316 | 322 |
317 RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat); | 323 RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat); |
318 }; | 324 }; |
319 | 325 |
320 } // namespace cricket | 326 } // namespace cricket |
321 | 327 |
322 #endif // TALK_SESSION_MEDIA_SRTPFILTER_H_ | 328 #endif // TALK_SESSION_MEDIA_SRTPFILTER_H_ |
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