One document matched: draft-jones-perc-private-media-framework-00.txt
Network Working Group P. Jones (Ed.)
Internet Draft N. Ismail
Intended status: Standards Track D. Benham
Expires: January 6, 2016 Cisco Systems
July 6, 2015
A Solution Framework for Private Media in Privacy Enhanced RTP
Conferencing
draft-jones-perc-private-media-framework-00
Abstract
This document describes a solution framework for ensuring that media
confidentiality and integrity are maintained end-to-end within the
context of a switched conferencing environment where media
distribution devices are not trusted with the end-to-end media
encryption keys. The solution aims to build upon existing security
mechanisms defined for the real-time transport protocol (RTP).
Status of this Memo
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 6, 2016.
Copyright Notice
Copyright (c) 2015 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
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to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction...................................................2
2. Requirements Language..........................................3
3. Private Media Trust Model......................................3
4. Solution Framework Overview....................................3
4.1. End-to-End Media Privacy..................................4
4.2. Hop-by-Hop Operations.....................................4
5. Private Media Packet Format....................................5
6. SRTP Cryptographic Context.....................................7
7. Cryptographic Operations.......................................8
7.1. Hop-by-Hop Authentication and Optional Encryption.........8
7.2. End-to-End Media Payload Encryption and Authentication....8
7.2.1. End-to-End Cryptographic Context Considerations......9
8. Key Exchange..................................................10
8.1. Session Signaling........................................10
8.2. Negotiating SRTP Protection Profiles and Key Exchange....12
8.2.1. Endpoint and KMF....................................12
8.2.2. MDD and KMF.........................................14
9. Changing Media Forwarded and EKT Field........................15
10. To-Do List...................................................15
10.1. What is needed to realize this Framework................15
10.2. Other Considerations for this Framework.................16
11. IANA Considerations..........................................16
12. Security Considerations......................................16
13. References...................................................16
13.1. Normative References....................................16
13.2. Informative References..................................17
14. Acknowledgments..............................................18
Authors' Addresses...............................................18
1. Introduction
Switched conferencing is an increasingly popular model for multimedia
conferences with multiple participants using a combination of audio,
video, text, and other media types. With this model, real-time media
flows from conference participants are not mixed, transcoded,
transrated, recomposed, or otherwise manipulated by a media
distribution device (MDD), as might be the case with a traditional
media server or multipoint control unit (MCU). Instead, media flows
transmitted by conference participants are simply forwarded by the
MDD to each of the other participants, often forwarding only a subset
of flows based on voice activity detection or other criteria. In
some instances, the switching MDDs may make limited modifications to
RTP [RFC3550] headers, for example, but the actual media content
(e.g., voice or video data) is unaltered.
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An advantage of switched conferencing is that MDDs can be deployed on
general-purpose computing hardware. This, in turn, means that it is
possible to deploy switching MDDs in virtualized environments,
including private and public clouds. Deploying conference resource in
a cloud environment might introduce a higher security risk. Whereas
traditional conference resources were usually deployed in private
networks that were protected, cloud-based conference resources might
be viewed as less secure since they are not always physically
controlled by those who use the hardware. Additionally, there are
usually several ports open to the public in cloud deployments, such
as for remote administration, and so on.
Recognizing the need to improve the way in which media
confidentiality is ensured, requirements for private media were
specified in [I.D-draft-jones-perc-private-media-reqts]. Attempting
to meet those requirements, this document defines a solution
framework wherein privacy is ensured by making it impossible for an
MDD to gain access to keys needed to decrypt or authenticate the
actual media content sent between conference participants. At the
same time, the framework allows for the switching MDD to modify
certain RTP headers; add, remove, encrypt, or decrypt RTP header
extensions; and encrypt and decrypt RTCP packets. The framework also
prevents replay attacks by authenticating each packet transmitted
between a given participant and the switching MDD by using a key that
is independent from the media encryption and authentication key(s)
and is unique to the participating endpoint and the switching MDD.
A goal of this framework is to meet the referenced requirements and
stated objectives by utilizing existing security procedures defined
for RTP with minimal extensions.
2. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119]
when they appear in ALL CAPS. These words may also appear in this
document in lower case as plain English words, absent their normative
meanings.
3. Private Media Trust Model
The private media trust model is specified in [I.D-draft-jones-perc-
private-media-reqts].
4. Solution Framework Overview
The purpose for this framework is to define a means through which
media privacy can be ensured when communicating within a switched
conferencing environment consisting of one or more centrally located
media distribution devices. This framework specifies the reuse of
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several technologies, including SRTP [RFC3711], EKT [I.D-draft-ietf-
avtcore-srtp-ekt], and DTLS-SRTP [RFC5764]. For the purposes of this
document, a conference refers to any session with two or more
participants - endpoints or other trusted devices - exchanging RTP
flows through media distribution devices.
4.1. End-to-End Media Privacy
This framework does not attempt to hide the fact that communication
between parties takes place. Rather, it only addresses the end-to-
end confidentiality and integrity of the actual media content.
To ensure the confidentiality and integrity of RTP media packets,
endpoints will utilize an EKT key - known to all conference
participants - to encrypt the SRTP key that is used to encrypt the
media (i.e., the RTP payload) via authenticated encryption.
Note that this EKT key may need to change from time-to-time during
the life of a conference, such as when a new participant joins or
leaves a conference. Dictating when a conference is to be re-keyed
is outside the scope of this document, but this framework does enable
re-keying of the conference.
Endpoints MUST maintain a list of SSRCs, track received sequence
number values relating to those SSRCs, and maintain associated SRTP
master keys for those SSRCs. All of this information SHOULD be
retained for some reasonable period of time and SHOULD be discarded
shortly after the EKT key for the conference is changed and upon
leaving the conference. However, following a change of the EKT key,
old key material SHOULD be retained long enough to ensure that late-
arriving or out-of-order packets can be successfully played.
4.2. Hop-by-Hop Operations
To ensure the integrity of transmitted media packets, this framework
requires that every packet be authenticated hop-by-hop. The
authentication key used for hop-by-hop authentication is derived from
an SRTP master key shared only on the respective hop between the
endpoint and the MDD to which it is attached. If MDDs are cascaded,
then there will also be an SRTP master key and derived authentication
key shared between the cascaded servers. Importantly, each of these
keys is distinct per hop and no two hops ever intentionally use the
same SRTP master key.
MDDs may find it necessary to change certain parts of the RTP packet
header, add or remove RTP header extensions, etc. By using hop-by-
hop authentication, the MDD is given liberty to change certain values
present in the RTP header, such as the payload type value.
If there is a desire to encrypt RTP header extensions, an encryption
key is derived from the hop-by-hop SRTP master key to encrypt header
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extensions as per [RFC6904]. This will give the switching MDD
visibility into header extensions, such as the one used to determine
audio level [RFC6464] of conference participants. Note that allowing
RTP header extensions to be encrypted requires that all hops decrypt
and re-encrypt any encrypted header extensions.
RTCP is optionally encrypted and mandatorily authenticated hop-by-hop
using the encryption and authentication keys derived from the SRTP
master key for the hop. This gives the switching MDD the flexibility
of either forwarding RTCP packets unchanged, transmit compound RTCP
packets, or to create RTCP packets to report statistics or for
conference control.
One of the reasons for performing hop-by-hop authentication is to
provide replay protection. If a media packet is replayed to the
switching MDD, it will be detected and rejected. Likewise, the
endpoint can detect replayed packets originally sent by the MDD.
Packets received by an endpoint that were originally sent to a
different endpoint will fail to pass authentication checks.
5. Private Media Packet Format
Since the RTP packet payload is encrypted and authenticated end-to-
end, extensions optionally encrypted hop-by-hop, and the entire RTP
packet is authenticated hop-by-hop, it may be useful to see the
entire RTP packet similarly to what is shown in [RFC3711].
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P|X| CC |M| PT | sequence number | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| timestamp | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| synchronization source (SSRC) identifier | |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| contributing source (CSRC) identifiers | |
| .... | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| RTP extension (OPTIONAL*) | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | payload ... | |
| | +-------------------------------+ |
| | | RTP padding | RTP pad count | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ Master Key Identifier (MKI) for End-to-End ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | EKT ciphertext ... | |
| | +-------------------------------+ |
| | | SRTP ROC (upper 16 bits) | |
| +-------------------------------+-------------------------------+ |
| | SRTP ROC (lower 16 bits) | Security Parameter Index |1| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTP MKI (OPTIONAL) for Hop-By-Hop ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag (MANDATORY) : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
+-- Authenticated Encryption Authenticated Portion --+
End-to-End using Hop-by-Hop Key
* Header extensions are optionally Encrypted Hop-by-Hop
Figure 1 - Private Media SRTP Packet with Full EKT Field
The rollover counter value is shown and transmitted as plaintext.
This is necessary since a switching MDD may not transmit media from a
"silent" participant's endpoint to others in the conference for a
long period of time. When media from that previously "silent"
participant is later forwarded to others, the receiving endpoint(s)
would not otherwise know the value of the rollover counter. Further,
this value is needed so that the correct authentication tag can be
generated hop-by-hop. Since the expected length of the EKT Field
might not be known to the MDD that is only authenticating packets,
the ROC field is placed as shown to ensure that its location can be
consistently determined.
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The EKT Field shown in Figure 1 is the "Full EKT Field". The "Short
EKT Field" may also be present in its place. When the short EKT
Field is used, the packet format looks like the one shown in Figure
2.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P|X| CC |M| PT | sequence number | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| timestamp | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| synchronization source (SSRC) identifier | |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| contributing source (CSRC) identifiers | |
| .... | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| RTP extension (OPTIONAL*) | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | payload ... | |
| | +-------------------------------+ |
| | | RTP padding | RTP pad count | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ Master Key Identifier (MKI) for End-to-End ~ |
| ~ +-------------------------------+ |
| ~ | Security Parameter Index |0| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTP MKI (OPTIONAL) for Hop-By-Hop ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag (MANDATORY) : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
+-- Authenticated Encryption Authenticated Portion --+
End-to-End using Hop-by-Hop Key
* Header extensions are optionally Encrypted Hop-by-Hop
Figure 2 - Private Media SRTP Packet with Short EKT Field
Note that the ROC field is absent when the short EKT Field is
transmitted. The assumption is that endpoints will transmit the Full
EKT Field regularly and frequently (e.g., every 100ms) to ensure that
media transmitted by a previously "silent" participant's endpoint can
be properly decrypted by other endpoints within a period of time that
is not noticeable to the human user.
6. SRTP Cryptographic Context
For any given media source identified by its SSRC, there is a single
SRTP cryptographic context as described in Section 3.2 of [RFC3711]
used in this framework.
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For end-to-end encryption, this framework extends the parameter set
of the cryptographic context by adding an identifier for the end-to-
end authenticated encryption algorithm. That parameter has
associated with it an EKT key (and associated EKT information, such
as master salt, key length, etc.), one or more SRTP master keys, and
as outlined in Section 3.2.1 of [RFC3711], other associated values
that relate to the master keys (e.g., master salt and key length
values).
For hop-by-hop encryption, the existing parameters in the SRTP
cryptographic context are used, including for the optional encryption
of RTP header extensions, authentication tag generation, etc.
7. Cryptographic Operations
7.1. Hop-by-Hop Authentication and Optional Encryption
For operations that occur hop-by-hop, the cryptographic transforms
defined in SRTP [RFC3711] (or other standardized transforms) may be
used in order optionally encrypt RTP header extensions, authenticate
the RTP packet, optionally encrypt the RTCP packet, and to
authenticate the RTCP packet.
The encryption and authentication of the RTP payload (media content)
itself is not a hop-by-hop operation, as explained in the next
section.
The procedures for optionally encrypting RTP header extensions is
define in [RFC6904] and MUST be used when encrypting header
extensions using the hop-by-hop SRTP master key to derive the k_he
and k_hs values.
The procedures for authenticating the RTP packet, optionally
encrypting the RTCP packet, and for authenticating the RTCP packet
shall follow the procedures defined in [RFC3711] using the hop-by-hop
SRTP master key and master salt to derive additional keys as
specified in that specification.
7.2. End-to-End Media Payload Encryption and Authentication
This section covers the encryption and authentication of the RTP
payload (i.e., media content) using the SRTP master key(s) created by
the endpoint and securely conveyed in the EKT Field using the EKT
key(s) shared with each of the endpoints in the conference.
This framework requires that the end-to-end cryptographic transforms
use authenticated encryption with associated data (AEAD) algorithms.
Specifically, the transforms defined in [I.D-draft-ietf-avtcore-srtp-
aes-gcm] are used as the mandatory transforms in this framework.
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The procedures followed to encrypt the payload are those described in
[I.D-draft-ietf-avtcore-srtp-aes-gcm], except that the associated
data used with those algorithms specified in Section 8.2 is redefined
as follows:
Associated Data: The version V (2 bits), padding flag P (1 bit),
CSRC count CC (4 bits), marker M (1 bit), the
sequence number (16 bits), timestamp (32 bits),
SSRC (32 bits), and optional contributing source
identifiers (CSRCs, 32 bits each).
The authentication tag for the end-to-end encrypted payload
immediately follows the encrypted payload in the defined packet
format (see section 5).
Note that RTP header extensions are not encrypted as a part of the
end-to-end operations. Rather, they are encrypted as a hop-by-hop
operation as explained in the previous section.
Only a part of the RTP packet is authenticated with the above
definition of "Associated Data" since packets are authenticated hop-
by-hop and there is a desire to allow switching MDDs to make changes
to certain parts of the RTP header. For these reasons, there is a
need for an authentication tag as defined in [RFC3711] to be placed
at the end of the RTP packet. This authentication tag is provided
via the hop-by-hop authentication operation as discussed in the
previous section. Note that this is also a deviation from what [I.D-
draft-ietf-avtcore-srtp-aes-gcm][DB1] recommends, but is necessary to
allow the switching MDD to make changes to certain fields that would
otherwise be protected.
7.2.1. End-to-End Cryptographic Context Considerations
7.2.1.1. Initialization Vector Formation
SRTP defines the following Initialization Vector (IV) as part of the
context for the AES Counter Mode cipher when encrypting RTP packets:
SRTP IV =
(SALT << 16) XOR (SSRC << 64) XOR (ROC << 32) XOR (SEQ << 16)
Following similar logic, [I.D-draft-ietf-avtcore-srtp-aes-gcm]
defines an Initialization Vector for encrypting RTP as follows:
SRTP IV =
SALT XOR (0x00 || 0x00 || SSRC ||ROC || SEQ)
Since this context includes and makes use of the SSRC, SEQ, and ROC,
these parameters must be preserved end-to-end for proper cipher
operation. In some RTP topologies, for example, a Selective
Forwarding Middlebox (SFM) with a common SSRC space, these parameters
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are preserved end-to-end because RTP middleboxes do not alter them.
In other RTP topologies, such as a Media Switching Mixer or a SFM
with separate SSRC spaces, the RTP middleboxes may need to alter them
for proper operation. When any of these parameters is altered, the
original parameters must still be preserved elsewhere in the packet
since they are essential parts of the cipher context. An RTP header
extension for End-to-End IV (EEIV) is defined for forwarding this
original context.
7.2.1.2. End-to-End IV (EEIV) RTP Header Extension
The End-to-End IV (EEIV) RTP Header Extension [RFC5285] conveys some
or all of the original end-to-end cipher context parameters: SSRC,
SEQ, and ROC. The extension has the following format:
EEIV = SSRC || SEQ
The extension MUST be added (if absent) by any RTP middlebox that
alters these parameters in the RTP header. It MUST NOT be added,
removed or altered if already present. Endpoints MAY add this
extension to operate with a RTP middlebox that can forward the
extension, but not add such an extension itself.
When an endpoint receives this extension in an SRTP packet, the
endpoint MUST use these values as the SSRC and RTP sequence number
when performing the authenticated decryption step as opposed to the
values found in the RTP header.
Note that he ROC does not need to be present in the header extension
since it is included in the payload as proposed in Section 5.
8. Key Exchange
Within this framework, there are various keys each endpoint needs:
those for end-to-end encryption/authentication and those for hop-by-
hop authentication, optional encryption of RTP header extensions,
SRTCP authentication, and optional SRTCP encryption. Likewise, the
MDD needs a hop-by-hop key when communicating with an endpoint or
cascaded conference server. The challenge is in securely exchanging
these keys between the appropriate entities.
To facilitate key exchange, this framework utilizes DTLS-SRTP and
procedures defined in EKT. This is explained further in the
following sub-sections.
8.1. Session Signaling
The session signaling protocol is not significant to this
specification, since the call processing functions are not assumed to
be trusted. Signaling might be via SIP [RFC3261] or a proprietary
signaling between a browser and a server, as examples. What is
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important is that the signaling convey, in some manner, the
fingerprint of the endpoint's certificate that will be used with
DTLS-SRTP. For the sake of providing a more concrete discussion, it
is assumed that SIP is used and SDP [RFC4566] conveys the fingerprint
information per [RFC5763].
The endpoint ("User Agent" in SIP terminology) will send an INVITE
message containing SDP for the media session along with the
endpoint's certificate fingerprint. This message or part thereof
MUST be cryptographically signed so as to prevent unauthorized,
undetectable modification of the fingerprint value, or the message
MUST be sent to a trusted element over a secure connection.
For this example, it is assumed that the endpoint sends a message to
a call processing function (e.g., a B2BUA) over a TLS connection.
The B2BUA might sign the message using the procedures described in
[RFC4474] for the benefit of forwarding the message to other
entities. It's important to note, however, that this does not lend
to the security of media, as the call processing function is not
assumed to be trusted.
An associated Key Management Function (KMF) needs to receive
information about the call and the endpoint(s). This might be
performed via an interface between the endpoint and the KMF, the call
processing function and the KMF, or it might be via a signaling
interface between the MDD and the KMF (see Figure 6 in [I.D-draft-
jones-perc-private-media-reqts]).
Regardless of the exact method, it is important that the endpoint's
certificate fingerprint and a participant identifier (a random value
created by the endpoint and provided to the KMF for each RTP session)
are securely conveyed to the KMF. The client certificate and
participant identifier will allow the KMF to associate the DTLS
connection to the specific endpoint and RTP session for the
conference.
Ultimately, a call is established to a conference and the endpoint
receives address information to which it may establish one or more
RTP sessions through to a MDD.
Call signaling going back to the endpoint might contain the
certificate fingerprint of the KMF that will process DTLS-SRTP
messages. Alternatively, the endpoint might already know the
certificate fingerprint. Whatever mechanism is employed, it is
extremely vital that the endpoint be able to fully trust the validity
of the fingerprint information for the KMF.
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8.2. Negotiating SRTP Protection Profiles and Key Exchange
8.2.1. Endpoint and KMF
There is a need for an SRTP master key and STRP master salt for hop-
by-hop authentication and optional encryption known to the endpoint
and the MDD. Additionally, there is a need to exchange an EKT master
key and EKT master salt for the end-to-end encryption of the media
content that is known to all participants in the conference, but not
known to the switching MDD.
To convey keys, the endpoint uses the procedures defined in [I.D-
draft-ietf-avtcore-srtp-ekt] for DTLS-SRTP over the media ports for
the RTP session. However, the switching MDD does not terminate the
DTLS signaling. Rather, DTLS packets received by the switching MDD
are forwarded to the KMF and vice versa. The figure below depicts
this.
+----------------------------+
+-----+ | Switching MDD |
| | | |
| KMF |<--------------->|<------------+ (Tunnels |
| | DTLS- | v DTLS-SRTP) |
+-----+ SRTP +----------------------------+
Tunnel ^
|
| DTLS-SRTP
|
v
+----------+
| Endpoint |
+----------+
Figure 3 - DTLS-SRTP Tunneled to KMF
Through this tunneled DTLS-SRTP exchange, an EKT master key and EKT
master salt are conveyed from the KMF to the endpoint, which the
endpoint will use when conveying SRTP keys and encrypt and
authenticate the media content in SRTP packets. The DTLS-SRTP
message exchanges between the switching MDD and KMF are encapsulated
in a second DTLS connection wherein the KMF also provides the MDD
with the hop-by-hop key material.
The KMF is described as a logical function in this document where the
functionality needed might be provided by one or more physical or
virtual entities. For example, there would obviously be a device
needed to terminate the DTLS-SRTP signaling. However, that device
may or may not be in possession of the EKT key used for the
conference. There DTLS-SRTP termination function might interface
with a Key Management Server (KMS), such as the one described in
[I.D-draft-abiggs-saag-key-management-service].
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{Editor's Note: a DTLS encapsulation protocol has been selected, but
has not been published in a separate draft. If there is no objection
to this approach, a proposal draft for the tunneling protocol will be
prepared.}
The endpoint does not transmit media encryption keys to the KMF. The
endpoint will follow the procedures specified in the EKT
specification to generate an SRTP master key and convey this
information to conference participants periodically (and anytime an
I-Frame is explicitly requested) via the "Full EKT Field."
This tunneling approach also needs an extension to EKT in order to
negotiate the SRTP Protection Profile used for end-to-end encryption
and authentication. The RECOMMENDED default protection profile is
AEAD_AES_128_GCM [I.D-draft-ietf-avtcore-srtp-aes-gcm].
The DTLS-SRTP procedures will result in the determination of an SRTP
master key and master salt, along with an SRTP Protection Profile.
This information is used for the hop-by-hop operations.
During the lifetime of a conference, the KMF MAY send a new EKT
message to endpoints providing a new EKT key to use from that point
forward, which might be desired when an endpoint leaves a conference
for example.
If a new endpoint joins a conference and does not support the same
SRTP Protection Profile in use, the KMF must initiate a new DTLS-SRTP
handshake with all conference participants to negotiate a new
security profile and to re-key the conference. This may cause some
disruption to conference. Therefore, it is recommended that only a
small number of protection profiles be required to implement by all
endpoints.
To help in understanding better the sequence of messages and the
relationship between the endpoint, MDD, and KMF, consider the
following figure:
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Endpoint KMF MDD
| | |
| External Signaling | |
| to exchange Cert and | |
| and participant ID | |
| ----------------------> | |
| | |
| DTLS connection | |
| -------------------------------------------------> | \
| | <======================= | /
| | DTLS Tunnel |
| | |
| For brevity, === is used for tunneled DTLS messages to KMF
| | |
| DTLS-SRTP and EKT | |
| ======================> | |
| (Participant ID, cert, | |
| security profiles, | |
| etc.) | |
| | |
| <=====================> | |
| SRTP Master key | -----------------------> |
| and salt determined | SRTP Master keys |
| for hop-by-hop | and salts conveyed |
| | for hop-by-hop |
| <====================== | (interface and |
| EKT key conveyed | endpoint/conference |
| for end-to-end | association TBD) |
| | |
| | |
Figure 4 - Key Exchange Procedure
Following the key exchange, the endpoint will be able to encrypt
media end-to-end and authenticate packets hop-by-hop. Likewise, the
conference server will be able to authenticate the received packet at
the hop, but will have no visibility into the encrypted media
content.
8.2.2. MDD and KMF
The DTLS tunnel between the MDD and the KMF used to encapsulate the
DTLS-SRTP signaling will also be used to convey the hop-by-hop
encryption keys, salt, and protection profile information. In this
way, no additional messages or interfaces are required in order for
the switching MDD to receive the required security parameters.
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9. Changing Media Forwarded and EKT Field
Endpoints transmit media to the MDD as they would to a traditional
conference server, except that media is encrypted and authenticated
with different keys as outlined in this framework. Each media source
within an RTP session has a distinct SSRC and endpoints work to
address SSRC collisions when they occur (see Section 8.2 of
[RFC3550]). From the endpoint's perspective, what is particularly
unique about the model described in this document is how the RTP
payload (media content) is encrypted and authenticated end-to-end,
while other security procedures are performed hop-by-hop.
To ensure a speedy decoder synchronization in receivers when
transitioning from forwarding one active speaker's media to the next,
a switching MDD will send a request for Full Intra-frame Request
(FIR) [RFC5104] (also known as a "video fast update" in [H.323]
systems) when a decision is made to switch active video flows. When
the endpoint receives this request, it would transmit the video frame
as requested and include with that initial packet the current "Full
EKT Field" so that recipients will be able to decrypt the media flow.
Additionally, a "Full EKT Field" should be transmitted about every
100ms to ensure that conference participants can decrypt the media
transmitted.
It is not possible to request a "Full EKT Field" for audio flows.
For this reason, it is RECOMMENDED that a "Full EKT Field" be
included in audio packets about every 100ms to smooth the transition
of the active speaker's audio forwarded by the server.
Endpoints SHOULD NOT include the "Full EKT Field" more frequently
than specified herein, rather opting for the "Short EKT Field" when
sending most packets to reduce the bandwidth consumed on the wire.
Endpoints MUST implement [RFC6464] in order for an MDD to determine
which endpoint(s) have an active speaker as no other method requiring
access to decrypted media can be used by an untrusted MDD.
10. To-Do List
10.1. What is needed to realize this Framework
- Endpoint must securely convey its certificate information to
the KMF so the KMF can recognize a valid endpoint.
- A means through EKT or another mechanism to negotiate the SRTP
security profiles for end-to-end encryption/authentication
(e.g., proposing to negotiate AEAD_AES_128_GCM for end-to-end
security)
- A means through EKT or another extension of sending the
participant identifier (the participant identifier could
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implicitly identify the conference) so the KMF will know which
keys to provide for a given conference and RTP sessions related
to that conference. Alternatively, this could be an element of
the tunneling protocol, wherein the MDD indicates the
associated identifiers.
- A change to EKT such that the ROC is transmitted in the clear,
with integrity check performed by XORing the ROC with the IV
used in AES Key Wrap
- A change to EKT to use MKI rather than ISN. This was proposed
by John Mattsson during IETF 92 to address security issues and
found to be extremely useful in easily managing multiple keys
over a period of time. Use of MKI could be avoided if when a
packet is received with a Full EKT Field, the key inside
replacing any previously received key for that SSRC. However,
in that case, the previous key would need to be retained for
some period of time to handle out-of-order packets.
- A means of conveying per-hop SRTP master key and salt
information to the switching MDD (which can be accomplished
using the DTLS-SRTP tunneling protocol)
10.2. Other Considerations for this Framework
- Investigate adding ability to enable one-way media from a non-
trusted device (e.g., announcements). While not specified as a
requirement, it was mentioned during a previous IETF meeting
and may be worth considering.
11. IANA Considerations
There are no IANA considerations for this document.
12. Security Considerations
[TBD]
13. References
13.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
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[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the
Secure Real-time Transport Protocol (SRTP)", RFC 5764,
May 2010.
[I.D-draft-ietf-avtcore-srtp-ekt]
Mattson, J., McGrew, D., Wing, D., and F. Andreasen,
"Encrypted Key Transport for Secure RTP", Work in
Progress, October 2014.
[RFC6904] J. Lennox, "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, December
2013.
[I.D-draft-ietf-avtcore-srtp-aes-gcm]
McGrew, D. and K. Igoe, "AES-GCM Authenticated Encryption
in Secure RTP (SRTP)", Work in Progress, June 2015.
[RFC5763] Fischl, J. Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, May 2010.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, December 2011.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[I.D-draft-abiggs-saag-key-management-service]
Biggs, A. and S. Cooley, "Key Management Service
Architecture", Work in Progress, July 2016.
13.2. Informative References
[I.D-draft-jones-perc-private-media-reqts]
Jones, P. et al., "Private Media Requirements in Privacy
Enhanced RTP Conferencing", Work in Progress, July 2015.
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[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[H.323] Recommendation ITU-T H.323, "Packet-based multimedia
communications systems", December 2009.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
14. Acknowledgments
The authors would like to thank Mo Zanaty and Christian Oien for
invaluable input on this document.
Authors' Addresses
Paul E. Jones
Cisco Systems, Inc.
7025 Kit Creek Rd.
Research Triangle Park, NC 27709
USA
Phone: +1 919 476 2048
Email: paulej@packetizer.com
Nermeen Ismail
Cisco Systems, Inc.
170 W Tasman Dr.
San Jose
USA
Email: nermeen@cisco.com
David Benham
Cisco Systems, Inc.
170 W Tasman Dr.
San Jose
USA
Email: dbenham@cisco.com
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