| /* |
| * 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 "pc/srtp_session.h" |
| |
| #include "media/base/rtp_utils.h" |
| #include "pc/external_hmac.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "third_party/libsrtp/include/srtp.h" |
| #include "third_party/libsrtp/include/srtp_priv.h" |
| |
| namespace cricket { |
| |
| // One more than the maximum libsrtp error code. Required by |
| // RTC_HISTOGRAM_ENUMERATION. Keep this in sync with srtp_error_status_t defined |
| // in srtp.h. |
| constexpr int kSrtpErrorCodeBoundary = 28; |
| |
| SrtpSession::SrtpSession() {} |
| |
| SrtpSession::~SrtpSession() { |
| if (session_) { |
| srtp_set_user_data(session_, nullptr); |
| srtp_dealloc(session_); |
| } |
| if (inited_) { |
| DecrementLibsrtpUsageCountAndMaybeDeinit(); |
| } |
| } |
| |
| bool SrtpSession::SetSend(int cs, |
| const uint8_t* key, |
| size_t len, |
| const std::vector<int>& extension_ids) { |
| return SetKey(ssrc_any_outbound, cs, key, len, extension_ids); |
| } |
| |
| bool SrtpSession::UpdateSend(int cs, |
| const uint8_t* key, |
| size_t len, |
| const std::vector<int>& extension_ids) { |
| return UpdateKey(ssrc_any_outbound, cs, key, len, extension_ids); |
| } |
| |
| bool SrtpSession::SetRecv(int cs, |
| const uint8_t* key, |
| size_t len, |
| const std::vector<int>& extension_ids) { |
| return SetKey(ssrc_any_inbound, cs, key, len, extension_ids); |
| } |
| |
| bool SrtpSession::UpdateRecv(int cs, |
| const uint8_t* key, |
| size_t len, |
| const std::vector<int>& extension_ids) { |
| return UpdateKey(ssrc_any_inbound, cs, key, len, extension_ids); |
| } |
| |
| bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!session_) { |
| RTC_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) { |
| RTC_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) { |
| RTC_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_.IsCurrent()); |
| if (!session_) { |
| RTC_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) { |
| RTC_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) { |
| RTC_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_.IsCurrent()); |
| if (!session_) { |
| RTC_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) { |
| // Limit the error logging to avoid excessive logs when there are lots of |
| // bad packets. |
| const int kFailureLogThrottleCount = 100; |
| if (decryption_failure_count_ % kFailureLogThrottleCount == 0) { |
| RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err |
| << ", previous failure count: " |
| << decryption_failure_count_; |
| } |
| ++decryption_failure_count_; |
| RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError", |
| static_cast<int>(err), kSrtpErrorCodeBoundary); |
| return false; |
| } |
| return true; |
| } |
| |
| bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!session_) { |
| RTC_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) { |
| RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err; |
| RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtcpUnprotectError", |
| static_cast<int>(err), kSrtpErrorCodeBoundary); |
| return false; |
| } |
| return true; |
| } |
| |
| bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| 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 (srtp_context && srtp_context->session_keys && |
| srtp_context->session_keys->rtp_auth) { |
| external_hmac = reinterpret_cast<ExternalHmacContext*>( |
| srtp_context->session_keys->rtp_auth->state); |
| } |
| |
| if (!external_hmac) { |
| RTC_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_.IsCurrent()); |
| 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, |
| const std::vector<int>& extension_ids) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| |
| 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 { |
| RTC_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. |
| RTC_NOTREACHED(); |
| RTC_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)) { |
| RTC_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 (!extension_ids.empty()) { |
| policy.enc_xtn_hdr = const_cast<int*>(&extension_ids[0]); |
| policy.enc_xtn_hdr_count = static_cast<int>(extension_ids.size()); |
| } |
| policy.next = nullptr; |
| |
| if (!session_) { |
| int err = srtp_create(&session_, &policy); |
| if (err != srtp_err_status_ok) { |
| session_ = nullptr; |
| RTC_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) { |
| RTC_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, |
| const std::vector<int>& extension_ids) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (session_) { |
| RTC_LOG(LS_ERROR) << "Failed to create SRTP session: " |
| "SRTP session already created"; |
| return false; |
| } |
| |
| // This is the first time we need to actually interact with libsrtp, so |
| // initialize it if needed. |
| if (IncrementLibsrtpUsageCountAndMaybeInit()) { |
| inited_ = true; |
| } else { |
| return false; |
| } |
| |
| return DoSetKey(type, cs, key, len, extension_ids); |
| } |
| |
| bool SrtpSession::UpdateKey(int type, |
| int cs, |
| const uint8_t* key, |
| size_t len, |
| const std::vector<int>& extension_ids) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!session_) { |
| RTC_LOG(LS_ERROR) << "Failed to update non-existing SRTP session"; |
| return false; |
| } |
| |
| return DoSetKey(type, cs, key, len, extension_ids); |
| } |
| |
| int g_libsrtp_usage_count = 0; |
| rtc::GlobalLockPod g_libsrtp_lock; |
| |
| // static |
| bool SrtpSession::IncrementLibsrtpUsageCountAndMaybeInit() { |
| rtc::GlobalLockScope ls(&g_libsrtp_lock); |
| |
| RTC_DCHECK_GE(g_libsrtp_usage_count, 0); |
| if (g_libsrtp_usage_count == 0) { |
| int err; |
| err = srtp_init(); |
| if (err != srtp_err_status_ok) { |
| RTC_LOG(LS_ERROR) << "Failed to init SRTP, err=" << err; |
| return false; |
| } |
| |
| err = srtp_install_event_handler(&SrtpSession::HandleEventThunk); |
| if (err != srtp_err_status_ok) { |
| RTC_LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err; |
| return false; |
| } |
| |
| err = external_crypto_init(); |
| if (err != srtp_err_status_ok) { |
| RTC_LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err; |
| return false; |
| } |
| } |
| ++g_libsrtp_usage_count; |
| return true; |
| } |
| |
| // static |
| void SrtpSession::DecrementLibsrtpUsageCountAndMaybeDeinit() { |
| rtc::GlobalLockScope ls(&g_libsrtp_lock); |
| |
| RTC_DCHECK_GE(g_libsrtp_usage_count, 1); |
| if (--g_libsrtp_usage_count == 0) { |
| int err = srtp_shutdown(); |
| if (err) { |
| RTC_LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err; |
| } |
| } |
| } |
| |
| void SrtpSession::HandleEvent(const srtp_event_data_t* ev) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| switch (ev->event) { |
| case event_ssrc_collision: |
| RTC_LOG(LS_INFO) << "SRTP event: SSRC collision"; |
| break; |
| case event_key_soft_limit: |
| RTC_LOG(LS_INFO) << "SRTP event: reached soft key usage limit"; |
| break; |
| case event_key_hard_limit: |
| RTC_LOG(LS_INFO) << "SRTP event: reached hard key usage limit"; |
| break; |
| case event_packet_index_limit: |
| RTC_LOG(LS_INFO) |
| << "SRTP event: reached hard packet limit (2^48 packets)"; |
| break; |
| default: |
| RTC_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 |
