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/*
* libjingle
* Copyright 2009, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_SESSION_PHONE_SRTPFILTER_H_
#define TALK_SESSION_PHONE_SRTPFILTER_H_
#include <list>
#include <map>
#include <string>
#include <vector>
#include "talk/base/basictypes.h"
#include "talk/base/scoped_ptr.h"
#include "talk/base/sigslotrepeater.h"
#include "talk/session/phone/cryptoparams.h"
#include "talk/p2p/base/sessiondescription.h"
// Forward declaration to avoid pulling in libsrtp headers here
struct srtp_event_data_t;
struct srtp_ctx_t;
typedef srtp_ctx_t* srtp_t;
struct srtp_policy_t;
namespace cricket {
// Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
// in applications (voice) where the additional bandwidth may be significant.
// A 80-bit HMAC is always used for SRTCP.
// 128-bit AES with 80-bit SHA-1 HMAC.
extern const char CS_AES_CM_128_HMAC_SHA1_80[];
// 128-bit AES with 32-bit SHA-1 HMAC.
extern const char CS_AES_CM_128_HMAC_SHA1_32[];
// Key is 128 bits and salt is 112 bits == 30 bytes. B64 bloat => 40 bytes.
extern const int SRTP_MASTER_KEY_BASE64_LEN;
class SrtpSession;
class SrtpStat;
void EnableSrtpDebugging();
// Class to transform SRTP to/from RTP.
// Initialize by calling SetSend with the local security params, then call
// SetRecv once the remote security params are received. At that point
// Protect/UnprotectRt(c)p can be called to encrypt/decrypt data.
// TODO: Figure out concurrency policy for SrtpFilter.
class SrtpFilter {
public:
enum Mode {
PROTECT,
UNPROTECT
};
enum Error {
ERROR_NONE,
ERROR_FAIL,
ERROR_AUTH,
ERROR_REPLAY,
};
SrtpFilter();
~SrtpFilter();
// Whether the filter is active (i.e. crypto has been properly negotiated).
bool IsActive() const;
// Indicates which crypto algorithms and keys were contained in the offer.
// offer_params should contain a list of available parameters to use, or none,
// if crypto is not desired. This must be called before SetAnswer.
bool SetOffer(const std::vector<CryptoParams>& offer_params,
ContentSource source);
// Indicates which crypto algorithms and keys were contained in the answer.
// answer_params should contain the negotiated parameters, which may be none,
// if crypto was not desired or could not be negotiated (and not required).
// This must be called after SetOffer. If crypto negotiation completes
// successfully, this will advance the filter to the active state.
bool SetAnswer(const std::vector<CryptoParams>& answer_params,
ContentSource source);
// Encrypts/signs an individual RTP/RTCP packet, in-place.
// If an HMAC is used, this will increase the packet size.
bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len);
// Decrypts/verifies an invidiual RTP/RTCP packet.
// If an HMAC is used, this will decrease the packet size.
bool UnprotectRtp(void* data, int in_len, int* out_len);
bool UnprotectRtcp(void* data, int in_len, int* out_len);
// Update the silent threshold (in ms) for signaling errors.
void set_signal_silent_time(uint32 signal_silent_time_in_ms);
sigslot::repeater3<uint32, Mode, Error> SignalSrtpError;
protected:
bool StoreParams(const std::vector<CryptoParams>& offer_params,
ContentSource source);
bool NegotiateParams(const std::vector<CryptoParams>& answer_params,
CryptoParams* selected_params);
bool ApplyParams(const CryptoParams& send_params,
const CryptoParams& recv_params);
bool ResetParams();
static bool ParseKeyParams(const std::string& params, uint8* key, int len);
private:
enum State { ST_INIT, ST_SENTOFFER, ST_RECEIVEDOFFER, ST_ACTIVE };
State state_;
std::vector<CryptoParams> offer_params_;
talk_base::scoped_ptr<SrtpSession> send_session_;
talk_base::scoped_ptr<SrtpSession> recv_session_;
};
// Class that wraps a libSRTP session.
class SrtpSession {
public:
SrtpSession();
~SrtpSession();
// Configures the session for sending data using the specified
// cipher-suite and key. Receiving must be done by a separate session.
bool SetSend(const std::string& cs, const uint8* key, int len);
// Configures the session for receiving data using the specified
// cipher-suite and key. Sending must be done by a separate session.
bool SetRecv(const std::string& cs, const uint8* key, int len);
// Encrypts/signs an individual RTP/RTCP packet, in-place.
// If an HMAC is used, this will increase the packet size.
bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len);
// Decrypts/verifies an invidiual RTP/RTCP packet.
// If an HMAC is used, this will decrease the packet size.
bool UnprotectRtp(void* data, int in_len, int* out_len);
bool UnprotectRtcp(void* data, int in_len, int* out_len);
// Update the silent threshold (in ms) for signaling errors.
void set_signal_silent_time(uint32 signal_silent_time_in_ms);
sigslot::repeater3<uint32, SrtpFilter::Mode, SrtpFilter::Error>
SignalSrtpError;
private:
bool SetKey(int type, const std::string& cs, const uint8* key, int len);
static bool Init();
void HandleEvent(const srtp_event_data_t* ev);
static void HandleEventThunk(srtp_event_data_t* ev);
srtp_t session_;
int rtp_auth_tag_len_;
int rtcp_auth_tag_len_;
talk_base::scoped_ptr<SrtpStat> srtp_stat_;
static bool inited_;
static std::list<SrtpSession*> sessions_;
int last_send_seq_num_;
DISALLOW_COPY_AND_ASSIGN(SrtpSession);
};
// Class that collects failures of SRTP.
class SrtpStat {
public:
SrtpStat();
// Report RTP protection results to the handler.
void AddProtectRtpResult(uint32 ssrc, int result);
// Report RTP unprotection results to the handler.
void AddUnprotectRtpResult(uint32 ssrc, int result);
// Report RTCP protection results to the handler.
void AddProtectRtcpResult(int result);
// Report RTCP unprotection results to the handler.
void AddUnprotectRtcpResult(int result);
// Get silent time (in ms) for SRTP statistics handler.
uint32 signal_silent_time() const { return signal_silent_time_; }
// Set silent time (in ms) for SRTP statistics handler.
void set_signal_silent_time(uint32 signal_silent_time) {
signal_silent_time_ = signal_silent_time;
}
// Sigslot for reporting errors.
sigslot::signal3<uint32, SrtpFilter::Mode, SrtpFilter::Error>
SignalSrtpError;
private:
// For each different ssrc and error, we collect statistics separately.
struct FailureKey {
FailureKey()
: ssrc(0),
mode(SrtpFilter::PROTECT),
error(SrtpFilter::ERROR_NONE) {
}
FailureKey(uint32 in_ssrc, SrtpFilter::Mode in_mode,
SrtpFilter::Error in_error)
: ssrc(in_ssrc),
mode(in_mode),
error(in_error) {
}
bool operator <(const FailureKey& key) const {
return ssrc < key.ssrc || mode < key.mode || error < key.error;
}
uint32 ssrc;
SrtpFilter::Mode mode;
SrtpFilter::Error error;
};
// For tracing conditions for signaling, currently we only use
// last_signal_time. Wrap this as a struct so that later on, if we need any
// other improvements, it will be easier.
struct FailureStat {
FailureStat()
: last_signal_time(0) {
}
FailureStat(uint32 in_last_signal_time)
: last_signal_time(in_last_signal_time) {
}
void Reset() {
last_signal_time = 0;
}
uint32 last_signal_time;
};
// Inspect SRTP result and signal error if needed.
void HandleSrtpResult(const FailureKey& key);
std::map<FailureKey, FailureStat> failures_;
// Threshold in ms to silent the signaling errors.
uint32 signal_silent_time_;
DISALLOW_COPY_AND_ASSIGN(SrtpStat);
};
} // namespace cricket
#endif // TALK_SESSION_PHONE_SRTPFILTER_H_