Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(228)

Unified Diff: webrtc/pc/srtpfilter.cc

Issue 2976443002: Move SrtpSession and tests to their own files. (Closed)
Patch Set: Created 3 years, 5 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 side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/pc/srtpfilter.h ('k') | webrtc/pc/srtpfilter_unittest.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/pc/srtpfilter.cc
diff --git a/webrtc/pc/srtpfilter.cc b/webrtc/pc/srtpfilter.cc
index a7634757d6b537d3f4b72fab31f3bac93ce4c142..dde84bc14eb7484a2a5016fdbf25caa23d047702 100644
--- a/webrtc/pc/srtpfilter.cc
+++ b/webrtc/pc/srtpfilter.cc
@@ -14,16 +14,13 @@
#include <algorithm>
-#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/pc/srtpsession.h"
#include "webrtc/rtc_base/base64.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/byteorder.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/sslstreamadapter.h"
#include "webrtc/rtc_base/stringencode.h"
#include "webrtc/rtc_base/timeutils.h"
@@ -484,394 +481,4 @@ bool SrtpFilter::ParseKeyParams(const std::string& key_params,
return true;
}
-///////////////////////////////////////////////////////////////////////////////
-// SrtpSession
-
-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
« no previous file with comments | « webrtc/pc/srtpfilter.h ('k') | webrtc/pc/srtpfilter_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698