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Side by Side Diff: webrtc/pc/srtpfilter.cc

Issue 2976443002: Move SrtpSession and tests to their own files. (Closed)
Patch Set: Created 3 years, 5 months ago
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1 /* 1 /*
2 * Copyright 2009 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2009 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "webrtc/pc/srtpfilter.h" 11 #include "webrtc/pc/srtpfilter.h"
12 12
13 #include <string.h> 13 #include <string.h>
14 14
15 #include <algorithm> 15 #include <algorithm>
16 16
17 #include "third_party/libsrtp/include/srtp.h"
18 #include "third_party/libsrtp/include/srtp_priv.h"
19 #include "webrtc/media/base/rtputils.h" 17 #include "webrtc/media/base/rtputils.h"
20 #include "webrtc/pc/externalhmac.h" 18 #include "webrtc/pc/srtpsession.h"
21 #include "webrtc/rtc_base/base64.h" 19 #include "webrtc/rtc_base/base64.h"
22 #include "webrtc/rtc_base/buffer.h" 20 #include "webrtc/rtc_base/buffer.h"
23 #include "webrtc/rtc_base/byteorder.h" 21 #include "webrtc/rtc_base/byteorder.h"
24 #include "webrtc/rtc_base/checks.h" 22 #include "webrtc/rtc_base/checks.h"
25 #include "webrtc/rtc_base/logging.h" 23 #include "webrtc/rtc_base/logging.h"
26 #include "webrtc/rtc_base/sslstreamadapter.h"
27 #include "webrtc/rtc_base/stringencode.h" 24 #include "webrtc/rtc_base/stringencode.h"
28 #include "webrtc/rtc_base/timeutils.h" 25 #include "webrtc/rtc_base/timeutils.h"
29 26
30 namespace cricket { 27 namespace cricket {
31 28
32 // NOTE: This is called from ChannelManager D'tor. 29 // NOTE: This is called from ChannelManager D'tor.
33 void ShutdownSrtp() { 30 void ShutdownSrtp() {
34 // If srtp_dealloc is not executed then this will clear all existing sessions. 31 // If srtp_dealloc is not executed then this will clear all existing sessions.
35 // This should be called when application is shutting down. 32 // This should be called when application is shutting down.
36 SrtpSession::Terminate(); 33 SrtpSession::Terminate();
(...skipping 440 matching lines...) Expand 10 before | Expand all | Expand 10 after
477 std::string key_b64(key_params.substr(7)), key_str; 474 std::string key_b64(key_params.substr(7)), key_str;
478 if (!rtc::Base64::Decode(key_b64, rtc::Base64::DO_STRICT, 475 if (!rtc::Base64::Decode(key_b64, rtc::Base64::DO_STRICT,
479 &key_str, nullptr) || key_str.size() != len) { 476 &key_str, nullptr) || key_str.size() != len) {
480 return false; 477 return false;
481 } 478 }
482 479
483 memcpy(key, key_str.c_str(), len); 480 memcpy(key, key_str.c_str(), len);
484 return true; 481 return true;
485 } 482 }
486 483
487 ///////////////////////////////////////////////////////////////////////////////
488 // SrtpSession
489
490 bool SrtpSession::inited_ = false;
491
492 // This lock protects SrtpSession::inited_.
493 rtc::GlobalLockPod SrtpSession::lock_;
494
495 SrtpSession::SrtpSession() {}
496
497 SrtpSession::~SrtpSession() {
498 if (session_) {
499 srtp_set_user_data(session_, nullptr);
500 srtp_dealloc(session_);
501 }
502 }
503
504 bool SrtpSession::SetSend(int cs, const uint8_t* key, size_t len) {
505 return SetKey(ssrc_any_outbound, cs, key, len);
506 }
507
508 bool SrtpSession::UpdateSend(int cs, const uint8_t* key, size_t len) {
509 return UpdateKey(ssrc_any_outbound, cs, key, len);
510 }
511
512 bool SrtpSession::SetRecv(int cs, const uint8_t* key, size_t len) {
513 return SetKey(ssrc_any_inbound, cs, key, len);
514 }
515
516 bool SrtpSession::UpdateRecv(int cs, const uint8_t* key, size_t len) {
517 return UpdateKey(ssrc_any_inbound, cs, key, len);
518 }
519
520 bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
521 RTC_DCHECK(thread_checker_.CalledOnValidThread());
522 if (!session_) {
523 LOG(LS_WARNING) << "Failed to protect SRTP packet: no SRTP Session";
524 return false;
525 }
526
527 int need_len = in_len + rtp_auth_tag_len_; // NOLINT
528 if (max_len < need_len) {
529 LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length "
530 << max_len << " is less than the needed " << need_len;
531 return false;
532 }
533
534 *out_len = in_len;
535 int err = srtp_protect(session_, p, out_len);
536 int seq_num;
537 GetRtpSeqNum(p, in_len, &seq_num);
538 if (err != srtp_err_status_ok) {
539 LOG(LS_WARNING) << "Failed to protect SRTP packet, seqnum="
540 << seq_num << ", err=" << err << ", last seqnum="
541 << last_send_seq_num_;
542 return false;
543 }
544 last_send_seq_num_ = seq_num;
545 return true;
546 }
547
548 bool SrtpSession::ProtectRtp(void* p,
549 int in_len,
550 int max_len,
551 int* out_len,
552 int64_t* index) {
553 if (!ProtectRtp(p, in_len, max_len, out_len)) {
554 return false;
555 }
556 return (index) ? GetSendStreamPacketIndex(p, in_len, index) : true;
557 }
558
559 bool SrtpSession::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) {
560 RTC_DCHECK(thread_checker_.CalledOnValidThread());
561 if (!session_) {
562 LOG(LS_WARNING) << "Failed to protect SRTCP packet: no SRTP Session";
563 return false;
564 }
565
566 int need_len = in_len + sizeof(uint32_t) + rtcp_auth_tag_len_; // NOLINT
567 if (max_len < need_len) {
568 LOG(LS_WARNING) << "Failed to protect SRTCP packet: The buffer length "
569 << max_len << " is less than the needed " << need_len;
570 return false;
571 }
572
573 *out_len = in_len;
574 int err = srtp_protect_rtcp(session_, p, out_len);
575 if (err != srtp_err_status_ok) {
576 LOG(LS_WARNING) << "Failed to protect SRTCP packet, err=" << err;
577 return false;
578 }
579 return true;
580 }
581
582 bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
583 RTC_DCHECK(thread_checker_.CalledOnValidThread());
584 if (!session_) {
585 LOG(LS_WARNING) << "Failed to unprotect SRTP packet: no SRTP Session";
586 return false;
587 }
588
589 *out_len = in_len;
590 int err = srtp_unprotect(session_, p, out_len);
591 if (err != srtp_err_status_ok) {
592 LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err;
593 return false;
594 }
595 return true;
596 }
597
598 bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
599 RTC_DCHECK(thread_checker_.CalledOnValidThread());
600 if (!session_) {
601 LOG(LS_WARNING) << "Failed to unprotect SRTCP packet: no SRTP Session";
602 return false;
603 }
604
605 *out_len = in_len;
606 int err = srtp_unprotect_rtcp(session_, p, out_len);
607 if (err != srtp_err_status_ok) {
608 LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err;
609 return false;
610 }
611 return true;
612 }
613
614 bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
615 RTC_DCHECK(thread_checker_.CalledOnValidThread());
616 RTC_DCHECK(IsExternalAuthActive());
617 if (!IsExternalAuthActive()) {
618 return false;
619 }
620
621 ExternalHmacContext* external_hmac = nullptr;
622 // stream_template will be the reference context for other streams.
623 // Let's use it for getting the keys.
624 srtp_stream_ctx_t* srtp_context = session_->stream_template;
625 #if defined(SRTP_MAX_MKI_LEN)
626 // libsrtp 2.1.0
627 if (srtp_context && srtp_context->session_keys &&
628 srtp_context->session_keys->rtp_auth) {
629 external_hmac = reinterpret_cast<ExternalHmacContext*>(
630 srtp_context->session_keys->rtp_auth->state);
631 }
632 #else
633 // libsrtp 2.0.0
634 // TODO(jbauch): Remove after switching to libsrtp 2.1.0
635 if (srtp_context && srtp_context->rtp_auth) {
636 external_hmac = reinterpret_cast<ExternalHmacContext*>(
637 srtp_context->rtp_auth->state);
638 }
639 #endif
640
641 if (!external_hmac) {
642 LOG(LS_ERROR) << "Failed to get auth keys from libsrtp!.";
643 return false;
644 }
645
646 *key = external_hmac->key;
647 *key_len = external_hmac->key_length;
648 *tag_len = rtp_auth_tag_len_;
649 return true;
650 }
651
652 int SrtpSession::GetSrtpOverhead() const {
653 return rtp_auth_tag_len_;
654 }
655
656 void SrtpSession::EnableExternalAuth() {
657 RTC_DCHECK(!session_);
658 external_auth_enabled_ = true;
659 }
660
661 bool SrtpSession::IsExternalAuthEnabled() const {
662 return external_auth_enabled_;
663 }
664
665 bool SrtpSession::IsExternalAuthActive() const {
666 return external_auth_active_;
667 }
668
669 bool SrtpSession::GetSendStreamPacketIndex(void* p,
670 int in_len,
671 int64_t* index) {
672 RTC_DCHECK(thread_checker_.CalledOnValidThread());
673 srtp_hdr_t* hdr = reinterpret_cast<srtp_hdr_t*>(p);
674 srtp_stream_ctx_t* stream = srtp_get_stream(session_, hdr->ssrc);
675 if (!stream) {
676 return false;
677 }
678
679 // Shift packet index, put into network byte order
680 *index = static_cast<int64_t>(
681 rtc::NetworkToHost64(
682 srtp_rdbx_get_packet_index(&stream->rtp_rdbx) << 16));
683 return true;
684 }
685
686
687 bool SrtpSession::DoSetKey(int type, int cs, const uint8_t* key, size_t len) {
688 RTC_DCHECK(thread_checker_.CalledOnValidThread());
689
690 srtp_policy_t policy;
691 memset(&policy, 0, sizeof(policy));
692 if (cs == rtc::SRTP_AES128_CM_SHA1_80) {
693 srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
694 srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
695 } else if (cs == rtc::SRTP_AES128_CM_SHA1_32) {
696 // RTP HMAC is shortened to 32 bits, but RTCP remains 80 bits.
697 srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
698 srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
699 } else if (cs == rtc::SRTP_AEAD_AES_128_GCM) {
700 srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp);
701 srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp);
702 } else if (cs == rtc::SRTP_AEAD_AES_256_GCM) {
703 srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp);
704 srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp);
705 } else {
706 LOG(LS_WARNING) << "Failed to " << (session_ ? "update" : "create")
707 << " SRTP session: unsupported cipher_suite " << cs;
708 return false;
709 }
710
711 int expected_key_len;
712 int expected_salt_len;
713 if (!rtc::GetSrtpKeyAndSaltLengths(cs, &expected_key_len,
714 &expected_salt_len)) {
715 // This should never happen.
716 LOG(LS_WARNING) << "Failed to " << (session_ ? "update" : "create")
717 << " SRTP session: unsupported cipher_suite without length information"
718 << cs;
719 return false;
720 }
721
722 if (!key ||
723 len != static_cast<size_t>(expected_key_len + expected_salt_len)) {
724 LOG(LS_WARNING) << "Failed to " << (session_ ? "update" : "create")
725 << " SRTP session: invalid key";
726 return false;
727 }
728
729 policy.ssrc.type = static_cast<srtp_ssrc_type_t>(type);
730 policy.ssrc.value = 0;
731 policy.key = const_cast<uint8_t*>(key);
732 // TODO(astor) parse window size from WSH session-param
733 policy.window_size = 1024;
734 policy.allow_repeat_tx = 1;
735 // If external authentication option is enabled, supply custom auth module
736 // id EXTERNAL_HMAC_SHA1 in the policy structure.
737 // We want to set this option only for rtp packets.
738 // By default policy structure is initialized to HMAC_SHA1.
739 // Enable external HMAC authentication only for outgoing streams and only
740 // for cipher suites that support it (i.e. only non-GCM cipher suites).
741 if (type == ssrc_any_outbound && IsExternalAuthEnabled() &&
742 !rtc::IsGcmCryptoSuite(cs)) {
743 policy.rtp.auth_type = EXTERNAL_HMAC_SHA1;
744 }
745 if (!encrypted_header_extension_ids_.empty()) {
746 policy.enc_xtn_hdr = const_cast<int*>(&encrypted_header_extension_ids_[0]);
747 policy.enc_xtn_hdr_count =
748 static_cast<int>(encrypted_header_extension_ids_.size());
749 }
750 policy.next = nullptr;
751
752 if (!session_) {
753 int err = srtp_create(&session_, &policy);
754 if (err != srtp_err_status_ok) {
755 session_ = nullptr;
756 LOG(LS_ERROR) << "Failed to create SRTP session, err=" << err;
757 return false;
758 }
759 srtp_set_user_data(session_, this);
760 } else {
761 int err = srtp_update(session_, &policy);
762 if (err != srtp_err_status_ok) {
763 LOG(LS_ERROR) << "Failed to update SRTP session, err=" << err;
764 return false;
765 }
766 }
767
768 rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
769 rtcp_auth_tag_len_ = policy.rtcp.auth_tag_len;
770 external_auth_active_ = (policy.rtp.auth_type == EXTERNAL_HMAC_SHA1);
771 return true;
772 }
773
774 bool SrtpSession::SetKey(int type, int cs, const uint8_t* key, size_t len) {
775 RTC_DCHECK(thread_checker_.CalledOnValidThread());
776 if (session_) {
777 LOG(LS_ERROR) << "Failed to create SRTP session: "
778 << "SRTP session already created";
779 return false;
780 }
781
782 if (!Init()) {
783 return false;
784 }
785
786 return DoSetKey(type, cs, key, len);
787 }
788
789 bool SrtpSession::UpdateKey(int type, int cs, const uint8_t* key, size_t len) {
790 RTC_DCHECK(thread_checker_.CalledOnValidThread());
791 if (!session_) {
792 LOG(LS_ERROR) << "Failed to update non-existing SRTP session";
793 return false;
794 }
795
796 return DoSetKey(type, cs, key, len);
797 }
798
799 void SrtpSession::SetEncryptedHeaderExtensionIds(
800 const std::vector<int>& encrypted_header_extension_ids) {
801 RTC_DCHECK(thread_checker_.CalledOnValidThread());
802 encrypted_header_extension_ids_ = encrypted_header_extension_ids;
803 }
804
805 bool SrtpSession::Init() {
806 rtc::GlobalLockScope ls(&lock_);
807
808 if (!inited_) {
809 int err;
810 err = srtp_init();
811 if (err != srtp_err_status_ok) {
812 LOG(LS_ERROR) << "Failed to init SRTP, err=" << err;
813 return false;
814 }
815
816 err = srtp_install_event_handler(&SrtpSession::HandleEventThunk);
817 if (err != srtp_err_status_ok) {
818 LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err;
819 return false;
820 }
821
822 err = external_crypto_init();
823 if (err != srtp_err_status_ok) {
824 LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err;
825 return false;
826 }
827 inited_ = true;
828 }
829
830 return true;
831 }
832
833 void SrtpSession::Terminate() {
834 rtc::GlobalLockScope ls(&lock_);
835
836 if (inited_) {
837 int err = srtp_shutdown();
838 if (err) {
839 LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err;
840 return;
841 }
842 inited_ = false;
843 }
844 }
845
846 void SrtpSession::HandleEvent(const srtp_event_data_t* ev) {
847 RTC_DCHECK(thread_checker_.CalledOnValidThread());
848 switch (ev->event) {
849 case event_ssrc_collision:
850 LOG(LS_INFO) << "SRTP event: SSRC collision";
851 break;
852 case event_key_soft_limit:
853 LOG(LS_INFO) << "SRTP event: reached soft key usage limit";
854 break;
855 case event_key_hard_limit:
856 LOG(LS_INFO) << "SRTP event: reached hard key usage limit";
857 break;
858 case event_packet_index_limit:
859 LOG(LS_INFO) << "SRTP event: reached hard packet limit (2^48 packets)";
860 break;
861 default:
862 LOG(LS_INFO) << "SRTP event: unknown " << ev->event;
863 break;
864 }
865 }
866
867 void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
868 // Callback will be executed from same thread that calls the "srtp_protect"
869 // and "srtp_unprotect" functions.
870 SrtpSession* session = static_cast<SrtpSession*>(
871 srtp_get_user_data(ev->session));
872 if (session) {
873 session->HandleEvent(ev);
874 }
875 }
876
877 } // namespace cricket 484 } // namespace cricket
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