Move SrtpSession and tests to their own files.

webrtc/pc/srtpsession.cc

Index: webrtc/pc/srtpsession.cc
diff --git a/webrtc/pc/srtpsession.cc b/webrtc/pc/srtpsession.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e8b29460a71d6e6eeb7702832c240a34de4a7cff
--- /dev/null
+++ b/webrtc/pc/srtpsession.cc
@@ -0,0 +1,408 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/pc/srtpsession.h"
+
+#include "third_party/libsrtp/include/srtp.h"
+#include "third_party/libsrtp/include/srtp_priv.h"
+#include "webrtc/media/base/rtputils.h"
+#include "webrtc/pc/externalhmac.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/sslstreamadapter.h"
+
+namespace cricket {
+
+bool SrtpSession::inited_ = false;
+
+// This lock protects SrtpSession::inited_.
+rtc::GlobalLockPod SrtpSession::lock_;
+
+SrtpSession::SrtpSession() {}
+
+SrtpSession::~SrtpSession() {
+  if (session_) {
+    srtp_set_user_data(session_, nullptr);
+    srtp_dealloc(session_);
+  }
+}
+
+bool SrtpSession::SetSend(int cs, const uint8_t* key, size_t len) {
+  return SetKey(ssrc_any_outbound, cs, key, len);
+}
+
+bool SrtpSession::UpdateSend(int cs, const uint8_t* key, size_t len) {
+  return UpdateKey(ssrc_any_outbound, cs, key, len);
+}
+
+bool SrtpSession::SetRecv(int cs, const uint8_t* key, size_t len) {
+  return SetKey(ssrc_any_inbound, cs, key, len);
+}
+
+bool SrtpSession::UpdateRecv(int cs, const uint8_t* key, size_t len) {
+  return UpdateKey(ssrc_any_inbound, cs, key, len);
+}
+
+bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!session_) {
+    LOG(LS_WARNING) << "Failed to protect SRTP packet: no SRTP Session";
+    return false;
+  }
+
+  int need_len = in_len + rtp_auth_tag_len_;  // NOLINT
+  if (max_len < need_len) {
+    LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length "
+                    << max_len << " is less than the needed " << need_len;
+    return false;
+  }
+
+  *out_len = in_len;
+  int err = srtp_protect(session_, p, out_len);
+  int seq_num;
+  GetRtpSeqNum(p, in_len, &seq_num);
+  if (err != srtp_err_status_ok) {
+    LOG(LS_WARNING) << "Failed to protect SRTP packet, seqnum=" << seq_num
+                    << ", err=" << err
+                    << ", last seqnum=" << last_send_seq_num_;
+    return false;
+  }
+  last_send_seq_num_ = seq_num;
+  return true;
+}
+
+bool SrtpSession::ProtectRtp(void* p,
+                             int in_len,
+                             int max_len,
+                             int* out_len,
+                             int64_t* index) {
+  if (!ProtectRtp(p, in_len, max_len, out_len)) {
+    return false;
+  }
+  return (index) ? GetSendStreamPacketIndex(p, in_len, index) : true;
+}
+
+bool SrtpSession::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!session_) {
+    LOG(LS_WARNING) << "Failed to protect SRTCP packet: no SRTP Session";
+    return false;
+  }
+
+  int need_len = in_len + sizeof(uint32_t) + rtcp_auth_tag_len_;  // NOLINT
+  if (max_len < need_len) {
+    LOG(LS_WARNING) << "Failed to protect SRTCP packet: The buffer length "
+                    << max_len << " is less than the needed " << need_len;
+    return false;
+  }
+
+  *out_len = in_len;
+  int err = srtp_protect_rtcp(session_, p, out_len);
+  if (err != srtp_err_status_ok) {
+    LOG(LS_WARNING) << "Failed to protect SRTCP packet, err=" << err;
+    return false;
+  }
+  return true;
+}
+
+bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!session_) {
+    LOG(LS_WARNING) << "Failed to unprotect SRTP packet: no SRTP Session";
+    return false;
+  }
+
+  *out_len = in_len;
+  int err = srtp_unprotect(session_, p, out_len);
+  if (err != srtp_err_status_ok) {
+    LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err;
+    return false;
+  }
+  return true;
+}
+
+bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!session_) {
+    LOG(LS_WARNING) << "Failed to unprotect SRTCP packet: no SRTP Session";
+    return false;
+  }
+
+  *out_len = in_len;
+  int err = srtp_unprotect_rtcp(session_, p, out_len);
+  if (err != srtp_err_status_ok) {
+    LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err;
+    return false;
+  }
+  return true;
+}
+
+bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(IsExternalAuthActive());
+  if (!IsExternalAuthActive()) {
+    return false;
+  }
+
+  ExternalHmacContext* external_hmac = nullptr;
+  // stream_template will be the reference context for other streams.
+  // Let's use it for getting the keys.
+  srtp_stream_ctx_t* srtp_context = session_->stream_template;
+#if defined(SRTP_MAX_MKI_LEN)
+  // libsrtp 2.1.0
+  if (srtp_context && srtp_context->session_keys &&
+      srtp_context->session_keys->rtp_auth) {
+    external_hmac = reinterpret_cast<ExternalHmacContext*>(
+        srtp_context->session_keys->rtp_auth->state);
+  }
+#else
+  // libsrtp 2.0.0
+  // TODO(jbauch): Remove after switching to libsrtp 2.1.0
+  if (srtp_context && srtp_context->rtp_auth) {
+    external_hmac =
+        reinterpret_cast<ExternalHmacContext*>(srtp_context->rtp_auth->state);
+  }
+#endif
+
+  if (!external_hmac) {
+    LOG(LS_ERROR) << "Failed to get auth keys from libsrtp!.";
+    return false;
+  }
+
+  *key = external_hmac->key;
+  *key_len = external_hmac->key_length;
+  *tag_len = rtp_auth_tag_len_;
+  return true;
+}
+
+int SrtpSession::GetSrtpOverhead() const {
+  return rtp_auth_tag_len_;
+}
+
+void SrtpSession::EnableExternalAuth() {
+  RTC_DCHECK(!session_);
+  external_auth_enabled_ = true;
+}
+
+bool SrtpSession::IsExternalAuthEnabled() const {
+  return external_auth_enabled_;
+}
+
+bool SrtpSession::IsExternalAuthActive() const {
+  return external_auth_active_;
+}
+
+bool SrtpSession::GetSendStreamPacketIndex(void* p,
+                                           int in_len,
+                                           int64_t* index) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  srtp_hdr_t* hdr = reinterpret_cast<srtp_hdr_t*>(p);
+  srtp_stream_ctx_t* stream = srtp_get_stream(session_, hdr->ssrc);
+  if (!stream) {
+    return false;
+  }
+
+  // Shift packet index, put into network byte order
+  *index = static_cast<int64_t>(rtc::NetworkToHost64(
+      srtp_rdbx_get_packet_index(&stream->rtp_rdbx) << 16));
+  return true;
+}
+
+bool SrtpSession::DoSetKey(int type, int cs, const uint8_t* key, size_t len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
+  srtp_policy_t policy;
+  memset(&policy, 0, sizeof(policy));
+  if (cs == rtc::SRTP_AES128_CM_SHA1_80) {
+    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
+    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
+  } else if (cs == rtc::SRTP_AES128_CM_SHA1_32) {
+    // RTP HMAC is shortened to 32 bits, but RTCP remains 80 bits.
+    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
+    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
+  } else if (cs == rtc::SRTP_AEAD_AES_128_GCM) {
+    srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp);
+    srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp);
+  } else if (cs == rtc::SRTP_AEAD_AES_256_GCM) {
+    srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp);
+    srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp);
+  } else {
+    LOG(LS_WARNING) << "Failed to " << (session_ ? "update" : "create")
+                    << " SRTP session: unsupported cipher_suite " << cs;
+    return false;
+  }
+
+  int expected_key_len;
+  int expected_salt_len;
+  if (!rtc::GetSrtpKeyAndSaltLengths(cs, &expected_key_len,
+                                     &expected_salt_len)) {
+    // This should never happen.
+    LOG(LS_WARNING)
+        << "Failed to " << (session_ ? "update" : "create")
+        << " SRTP session: unsupported cipher_suite without length information"
+        << cs;
+    return false;
+  }
+
+  if (!key ||
+      len != static_cast<size_t>(expected_key_len + expected_salt_len)) {
+    LOG(LS_WARNING) << "Failed to " << (session_ ? "update" : "create")
+                    << " SRTP session: invalid key";
+    return false;
+  }
+
+  policy.ssrc.type = static_cast<srtp_ssrc_type_t>(type);
+  policy.ssrc.value = 0;
+  policy.key = const_cast<uint8_t*>(key);
+  // TODO(astor) parse window size from WSH session-param
+  policy.window_size = 1024;
+  policy.allow_repeat_tx = 1;
+  // If external authentication option is enabled, supply custom auth module
+  // id EXTERNAL_HMAC_SHA1 in the policy structure.
+  // We want to set this option only for rtp packets.
+  // By default policy structure is initialized to HMAC_SHA1.
+  // Enable external HMAC authentication only for outgoing streams and only
+  // for cipher suites that support it (i.e. only non-GCM cipher suites).
+  if (type == ssrc_any_outbound && IsExternalAuthEnabled() &&
+      !rtc::IsGcmCryptoSuite(cs)) {
+    policy.rtp.auth_type = EXTERNAL_HMAC_SHA1;
+  }
+  if (!encrypted_header_extension_ids_.empty()) {
+    policy.enc_xtn_hdr = const_cast<int*>(&encrypted_header_extension_ids_[0]);
+    policy.enc_xtn_hdr_count =
+        static_cast<int>(encrypted_header_extension_ids_.size());
+  }
+  policy.next = nullptr;
+
+  if (!session_) {
+    int err = srtp_create(&session_, &policy);
+    if (err != srtp_err_status_ok) {
+      session_ = nullptr;
+      LOG(LS_ERROR) << "Failed to create SRTP session, err=" << err;
+      return false;
+    }
+    srtp_set_user_data(session_, this);
+  } else {
+    int err = srtp_update(session_, &policy);
+    if (err != srtp_err_status_ok) {
+      LOG(LS_ERROR) << "Failed to update SRTP session, err=" << err;
+      return false;
+    }
+  }
+
+  rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
+  rtcp_auth_tag_len_ = policy.rtcp.auth_tag_len;
+  external_auth_active_ = (policy.rtp.auth_type == EXTERNAL_HMAC_SHA1);
+  return true;
+}
+
+bool SrtpSession::SetKey(int type, int cs, const uint8_t* key, size_t len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (session_) {
+    LOG(LS_ERROR) << "Failed to create SRTP session: "
+                  << "SRTP session already created";
+    return false;
+  }
+
+  if (!Init()) {
+    return false;
+  }
+
+  return DoSetKey(type, cs, key, len);
+}
+
+bool SrtpSession::UpdateKey(int type, int cs, const uint8_t* key, size_t len) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!session_) {
+    LOG(LS_ERROR) << "Failed to update non-existing SRTP session";
+    return false;
+  }
+
+  return DoSetKey(type, cs, key, len);
+}
+
+void SrtpSession::SetEncryptedHeaderExtensionIds(
+    const std::vector<int>& encrypted_header_extension_ids) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  encrypted_header_extension_ids_ = encrypted_header_extension_ids;
+}
+
+bool SrtpSession::Init() {
+  rtc::GlobalLockScope ls(&lock_);
+
+  if (!inited_) {
+    int err;
+    err = srtp_init();
+    if (err != srtp_err_status_ok) {
+      LOG(LS_ERROR) << "Failed to init SRTP, err=" << err;
+      return false;
+    }
+
+    err = srtp_install_event_handler(&SrtpSession::HandleEventThunk);
+    if (err != srtp_err_status_ok) {
+      LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err;
+      return false;
+    }
+
+    err = external_crypto_init();
+    if (err != srtp_err_status_ok) {
+      LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err;
+      return false;
+    }
+    inited_ = true;
+  }
+
+  return true;
+}
+
+void SrtpSession::Terminate() {
+  rtc::GlobalLockScope ls(&lock_);
+
+  if (inited_) {
+    int err = srtp_shutdown();
+    if (err) {
+      LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err;
+      return;
+    }
+    inited_ = false;
+  }
+}
+
+void SrtpSession::HandleEvent(const srtp_event_data_t* ev) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  switch (ev->event) {
+    case event_ssrc_collision:
+      LOG(LS_INFO) << "SRTP event: SSRC collision";
+      break;
+    case event_key_soft_limit:
+      LOG(LS_INFO) << "SRTP event: reached soft key usage limit";
+      break;
+    case event_key_hard_limit:
+      LOG(LS_INFO) << "SRTP event: reached hard key usage limit";
+      break;
+    case event_packet_index_limit:
+      LOG(LS_INFO) << "SRTP event: reached hard packet limit (2^48 packets)";
+      break;
+    default:
+      LOG(LS_INFO) << "SRTP event: unknown " << ev->event;
+      break;
+  }
+}
+
+void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
+  // Callback will be executed from same thread that calls the "srtp_protect"
+  // and "srtp_unprotect" functions.
+  SrtpSession* session =
+      static_cast<SrtpSession*>(srtp_get_user_data(ev->session));
+  if (session) {
+    session->HandleEvent(ev);
+  }
+}
+
+}  // namespace cricket