Double Encryption Procedures for the Secure Real-Time Transport Protocol (SRTP)
RFC 8723

Document Type RFC - Proposed Standard (April 2020; No errata)
Last updated 2020-04-30
Replaces draft-jennings-perc-double
Stream IETF
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Stream WG state Submitted to IESG for Publication
Document shepherd Suhas Nandakumar
Shepherd write-up Show (last changed 2018-05-03)
IESG IESG state RFC 8723 (Proposed Standard)
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Send notices to Suhas Nandakumar <suhasietf@gmail.com>
IANA IANA review state Version Changed - Review Needed
IANA action state RFC-Ed-Ack


Internet Engineering Task Force (IETF)                       C. Jennings
Request for Comments: 8723                                      P. Jones
Category: Standards Track                                      R. Barnes
ISSN: 2070-1721                                            Cisco Systems
                                                              A.B. Roach
                                                                 Mozilla
                                                              April 2020

Double Encryption Procedures for the Secure Real-Time Transport Protocol
                                 (SRTP)

Abstract

   In some conferencing scenarios, it is desirable for an intermediary
   to be able to manipulate some parameters in Real-time Transport
   Protocol (RTP) packets, while still providing strong end-to-end
   security guarantees.  This document defines a cryptographic transform
   for the Secure Real-time Transport Protocol (SRTP) that uses two
   separate but related cryptographic operations to provide hop-by-hop
   and end-to-end security guarantees.  Both the end-to-end and hop-by-
   hop cryptographic algorithms can utilize an authenticated encryption
   with associated data (AEAD) algorithm or take advantage of future
   SRTP transforms with different properties.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   https://www.rfc-editor.org/info/rfc8723.

Copyright Notice

   Copyright (c) 2020 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (https://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction
   2.  Terminology
   3.  Cryptographic Context
     3.1.  Key Derivation
   4.  Original Header Block
   5.  RTP Operations
     5.1.  Encrypting a Packet
     5.2.  Relaying a Packet
     5.3.  Decrypting a Packet
   6.  RTCP Operations
   7.  Use with Other RTP Mechanisms
     7.1.  RTP Retransmission (RTX)
     7.2.  Redundant Audio Data (RED)
     7.3.  Forward Error Correction (FEC)
     7.4.  DTMF
   8.  Recommended Inner and Outer Cryptographic Algorithms
   9.  Security Considerations
   10. IANA Considerations
     10.1.  DTLS-SRTP
   11. References
     11.1.  Normative References
     11.2.  Informative References
   Appendix A.  Encryption Overview
   Acknowledgments
   Authors' Addresses

1.  Introduction

   Cloud conferencing systems that are based on switched conferencing
   have a central Media Distributor (MD) device that receives media from
   endpoints and distributes it to other endpoints, but does not need to
   interpret or change the media content.  For these systems, it is
   desirable to have one cryptographic key that enables encryption and
   authentication of the media end-to-end while still allowing certain
   information in the header of an RTP packet to be changed by the MD.
   At the same time, a separate cryptographic key provides integrity and
   optional confidentiality for the media flowing between the MD and the
   endpoints.  The framework document [PRIVATE-MEDIA-FRAMEWORK]
   describes this concept in more detail.

   This specification defines a transform for SRTP that uses 1) the AES
   Galois/Counter Mode (AES-GCM) algorithm [RFC7714] to provide
   encryption and integrity for an RTP packet for the end-to-end
   cryptographic key and 2) a hop-by-hop cryptographic encryption and
   integrity between the endpoint and the MD.  The MD decrypts and
   checks integrity of the hop-by-hop security.  The MD MAY change some
   of the RTP header information that would impact the end-to-end
   integrity.  In that case, the original value of any RTP header field
   that is changed is included in an "Original Header Block" that is
   added to the packet.  The new RTP packet is encrypted with the hop-
   by-hop cryptographic algorithm before it is sent.  The receiving
   endpoint decrypts and checks integrity using the hop-by-hop
   cryptographic algorithm and then replaces any parameters the MD
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