Internet DRAFT - draft-jones-avtcore-private-media-framework
draft-jones-avtcore-private-media-framework
Network Working Group P. Jones (Ed.)
Internet Draft N. Ismail
Intended status: Standards Track D. Benham
Expires: September 7, 2015 Cisco Systems
March 7, 2015
A Solution Framework for Private Media in a Switched Conferencing
draft-jones-avtcore-private-media-framework-01
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 the switching
conference server is not entrusted with the media encryption keys.
The solution aims to build upon existing security mechanisms defined
for the real-time transport protocol (RTP).
Status of this Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on September 7, 2015.
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Table of Contents
1. Introduction...................................................2
2. Requirements Language..........................................3
3. Private Media Trust Model......................................3
4. Solution Framework Overview....................................3
4.1. Switching Conference Server Behavior......................4
4.2. End-to-End Media Privacy..................................5
4.3. Hop-by-Hop Operations.....................................6
5. Media Packet Format............................................6
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
8. Key Exchange...................................................9
8.1. Session Signaling.........................................9
8.2. Negotiating SRTP Protection Profiles and Key Exchange....11
8.2.1. Endpoint and KMF....................................11
8.2.2. Switching Conference Server and KMF.................13
9. Changing Media Forwarded and EKT Field........................14
10. IANA Considerations..........................................14
11. Security Considerations......................................14
12. References...................................................15
12.1. Normative References....................................15
12.2. Informative References..................................16
13. Acknowledgments..............................................16
Authors' Addresses...............................................17
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 on the conference
server, as might be the case with a traditional multipoint control
unit (MCU). Instead, media flows transmitted by conference
participants are simply forwarded by the switching conference server
to each of the other participants, selectively forwarding flows based
on voice activity detection or other criteria. In some instances,
the switching conference server may make limited modifications to RTP
[RFC3550] headers, for example, but the actual media content (e.g.,
voice or video data) is unaltered.
An advantage of switched conferencing is that conference servers can
be deployed on general-purpose computing hardware, as there is no
need for the specialized hardware required to manipulate media flows
that one finds on a traditional hardware MCU. This, in turn, means
that it is possible to deploy switching conference servers in
virtualized environments, including private and public clouds.
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However, deploying conference resource in a cloud environment may
introduce a higher security risk. Whereas traditional conference
servers were usually deployed in private networks that were protected
from public access by firewalls, 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, most
significantly being ports where the administrator can log in to make
configuration changes, install software updates, 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-avtcore-private-media-reqts].
Attempting to meet those requirements, this document defines a
solution framework wherein privacy is ensured by making it impossible
for a switching conference server 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 conference server 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 conference server 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 conference
server.
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-
avtcore-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. This framework specifies the re-use of
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several technologies, including SRTP [RFC3711], EKT [I.D-draft-ietf-
avtcore-srtp-ekt], and DTLS-SRTP [RFC5764].
4.1. Switching Conference Server Behavior
Before going into the specifics of how media privacy is ensured,
first consider Figure 1 below depicting the behavior of a switching
conferencing server forwarding media between participants.
+--------------------+
+---+ --{A}--> | | <-{C}--- +---+
| A | <-{B}--- |Switching Conference| --{A}--> | C |
| | <-{C}--- | Server | --{B}--> | |
+---+ <-{D}--- | | --{D}--> +---+
| Packet |
+---+ --{B}--> | Authentication | <-{D}--- +---+
| B | <-{A}--- | | --{A}--> | D |
| | <-{C}--- | | --{B}--> | |
+---+ <-{D}--- | Media Privacy | --{C}--> +---+
+--------------------+
Figure 1 - Switching Conference Server
In the above figure, each of the participating endpoints sends media
to the switching conference server where each flow is protected using
the security mechanisms that will be discussed later. Each endpoint
then receives media from each of the other participants in the
conference. Importantly, the switching conference server is unable
to decrypt the media content or modify media content without being
detected by receiving endpoints.
The framework does not require, however, for each of the participant
flows to be transmitted to every other endpoint in the conference.
In many situations, the switching conference server will transmit
only a subset of the media flows to each participant. This might be
to restrict the bandwidth usage, provide a primary video flow and
thumbnail flows to single-screen video endpoints, etc.
The following two diagrams and corresponding explanatory text are for
illustrative purposes to describe one possible operational mode.
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+--------------------+
+---+ --{A}--> | | <--{C}-- +---+
| A | |Switching Conference| | C |*
+---+ <-{C}--- | Server | ---{A}-> +---+
| |
+---+ --{B}--> | | <--{D}-- +---+
| B | | | | D |
+---+ <-{C}--- | | ---{C}-> +---+
+--------------------+
Figure 2 - Endpoint "C" is the Active Speaker
As depicted in Figure 2, each of the endpoints in the conference is
receiving a single flow. In particular, all but one endpoints are
receiving media flows from endpoint "C", the current active speaker.
Endpoint "C" is receiving media from endpoint "A", the former active
speaker.
+--------------------+
+---+ --{A}--> | | <--{C}-- +---+
| A | |Switching Conference| | C |
+---+ <-{B}--- | Server | ---{B}-> +---+
| |
+---+ --{B}--> | | <--{D}-- +---+
*| B | | | | D |
+---+ <-{C}--- | | ---{B}-> +---+
+--------------------+
Figure 3 - Endpoint "B" is the Active Speaker
When the active speaker transitions, so do the video flows. As
depicted in Figure 3, the active speaker transitions from "C" to "B".
Now, each of the endpoints receives a copy of the media flows from
"B", while "B" receives the media flow from "C", the former active
speaker.
How many flows and what type of flows a switching conference server
transmits to a receiving endpoint are outside the scope of this
document.
4.2. End-to-End Media Privacy
To ensure the confidentiality of RTP media packets, endpoints utilize
EKT keys known to conference participants to encrypt the media
content of the RTP packet (i.e., the RTP payload) using authenticated
encryption. These keys may change from time-to-time for various
reasons, such as when a new conference participant joins a conference
or when a conference participant leaves a conference. When it is
decided that a conference is to be re-keyed is outside the scope of
this document, but it is important that an unstrusted switching
conference server is never given access to those keys.
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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.
4.3. Hop-by-Hop Operations
To ensure the integrity of transmitted media packets, this framework
requires that every packet be authenticated. While media is both
encrypted and authenticated end-to-end, RTP packets are also
authenticated hop-by-hop. The authentication key used for hop-by-hop
authentication is derived from the SRTP master key shared only on the
respective hop. If conference servers are cascaded, then there will
also be SRTP master keys and derived authentication keys 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.
It is expected that the conference servers 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
switching media server 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
extensions as per [RFC6904]. This will give the switching conference
server 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 conference server
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 conference server, it will be detected. Likewise, the
endpoint can detect replayed packets originally sent by the media
server. Packets received by an endpoint that were originally sent to
a different endpoint will fail to pass authentication checks.
5. 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 | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| | SRTP ROC | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | EKT ciphertext ... | |
| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | | Security Parameter Index |1| |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| : authentication tag (MANDATORY) : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
+-- Authenticated Encryption Authenticated Portion --+
End-to-End using Hop-by-Hop Key
* Header extensions are optionally Encrypted Hop-by-Hop
Figure 4 - Private Media SRTP Packet
The rollover counter value is shown and transmitted as plaintext.
This is necessary since a switching conference server may not
transmit media from one "silent" participant to another participant
in the conference for a long period of time. When media from that
"silent" participant is later sent to that other participant, the
receiving participant 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.
The EKT field shown in Figure 4 is the "Full EKT Field". The "Short
EKT Field" may also be present in its place.
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. However, this framework extends the
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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 SRTP master key, and as outlined
in Section 3.2.1, other associated values that relate to the master
key (e.g., master salt and key length values). For AES-CCM, there
will also be an associated "Tag_Size_Flag" value (see [I.D-draft-
ietf-avtcore-srtp-aes-gcm]).
The existing parameters in the SRTP cryptographic context are used
for hop-by-hop operations, including 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) derived
from the EKT Key(s) by the endpoints communicating in a switched
conferencing environment.
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 default transforms in this framework.
The procedures followed to encrypt the payload are those described in
[I.D-draft-ietf-avtcore-srtp-aes-gcm], except that the associated
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data used with those algorithms specified in Section 9.2 is redefined
as follows:
Associated Data: The version V (2 bits), padding flag P (1 bit),
the sequence number (16 bits), timestamp (32
bits), and SSRC (32 bits).
The authentication tag for the end-to-end encrypted payload
immediately follows the encrypted payload in the packet format.
Note that RTP header extensions are not encrypted as a part of the
end-to-end function. 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 conference servers 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] recommends, but is necessary to
allow the switching conference server to make changes to certain
fields that would otherwise be protected.
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 encryptions,
SRTCP authentication, and optional SRTCP encryption. Likewise, the
switching conference server needs a hop-by-hop key when communicating
with an endpoint or cascaded conference server. The challenge is in
securely exchanging these keys to the appropriate entities.
To facilitate key exchange, we utilize DTLS-SRTP and procedures
defined in EKT. We will elaborate on this 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 untrusted.
Signaling might be via SIP [RFC3261] or a proprietary signaling
between a browser and a server, as examples. What is 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, we will assume SIP is
used and SDP [RFC4566] conveys the fingerprint information as per
[RFC5763].
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The endpoint ("User Agent" in SIP terminology) will send an INVITE
message containing SDP for the media session along with fingerprints.
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, we will assume 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, including the switching conference server. It's important
to note, however, that this does not lend to the security of media,
as the call processing function is not trusted.
The Key Management Function (KMF) needs to receive information about
the call. 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 switching conference
server and the KMF (see Error! Reference source not found.).
Regardless, 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. The endpoint
to KMF information exchange is outside the scope of this document.
Ultimately, a call is established on the switching conference server
and the endpoint receives address information to which it may
establish one or more RTP sessions.
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.
[Editor's Note: How would an endpoint that is outside an enterprise
domain (e.g., an associate at another company) be able to interact
with the enterprise KMF? It might be necessary to have a trusted
call processing entity that signs messages that the foreign endpoint
can validate so that it knows that it can trust the certificate
fingerprint of 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 conference server. 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 conference servers.
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 conference server does not
terminate the DTLS signaling. Rather, DTLS packets received by the
conference server are forwarded to the KMF and vice versa. The
figure below depicts this.
Conference
+-----+ Server / KMF +-----------------------------+
| | Interface | Switching Conference Server |
| |<--------------->| |
| | | |
| KMF |<--------------->|<-------------+ (Tunnels |
| | DTLS- | v DTLS-SRTP) |
+-----+ SRTP +-----------------------------+
Tunnel ^
| DTLS-SRTP
|
v
+----------+
| Endpoint |
+----------+
Figure 5 - 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 deriving SRTP keys and encrypt and
authenticate the media content in SRTP packets. 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". [Editor's note: we are proposing
changes to the EKT draft that will include the ROC separated from the
EKT Ciphertext. Additionally, we need a mechanism to negotiate SRTP
Protection Profiles for the end-to-end encryption/authentication.
This might be an extension to EKT, a new extension, or even an
application-layer exchange over the DTLS connection to the KMF.]
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This framework also calls for the extension of 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. [Editor's
note: We could use DTLS-SRTP only to negotiate the SRTP Protection
Profiles and then introduce a new extension to allow the KMF to send
out the hop-by-hop key and salt to both the endpoint and conference
server. Open to alternative suggestions from the workgroup.]
During the lifetime of the conference, conference participants may
come and go. During those events, the KMF will send a new EKT
message to clients providing a new EKT key to use from that point
forward.
If a new participant does not support the same SRTP Protection
Profile in use by the conference, 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 we
select a small number of protection profiles that must be implemented
by all endpoints.
Summary of what we need to realize this framework:
- Endpoint must securely convey its certificate information to
the KMF and negotiate a participant identifier (e.g., a UUID
securely conveyed, but need not be encrypted) before a
connection to the conference server is attempted.
- A means through EKT or another extension to negotiate the SRTP
security profiles for end-to-end encryption/authentication
- A means through EKT or another extension of sending the
participant identifier (the participant identifier could
implicitly identify the conference)
- 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 means of conveying per-hop SRTP master key and salt
information to the switching conference server
To help in understanding better the sequence of messages, consider
the following figure:
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Conference
Endpoint KMF Server
| | |
| External Signaling | |
| To exchange Cert and | |
| and participant ID | |
| ----------------------> | |
| | |
| DTLS connection | |
| -------------------------------------------------> | \
| | <======================= | /
| | DTLS Tunnel |
| | |
| For brevity, we use === 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 6 - 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. Switching Conference Server and KMF
[Editor's Note: there must be an interface between the switching
conference server and the KMF so that cipher suites and key
information can be conveyed for each participant in each conference
for hop-by-hop operations. This interface is out of scope for this
document.]
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9. Changing Media Forwarded and EKT Field
Endpoints transmit media to the switching conference server as they
would in a traditional conference, 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. 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 conference server 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.
A switching conference server may forward a single audio and video
flow to a receiver, or it may forward multiple flows. The number of
media flows very much depends on the capabilities of the receiving
device. How the number of media flows to forward is determined or
negotiated is outside the scope of this document.
To aid in determining when to transition the active speaker's audio
or video, endpoints MUST implement [RFC6464] in order to provide a
hint to the switching media server as to which endpoint should be
designated as the one of the active speaker(s).
10. IANA Considerations
There are no IANA considerations for this document.
11. Security Considerations
[TBD]
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12. References
12.1. 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.
[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 and AES-CCM
Authenticated Encryption in Secure RTP (SRTP)", Work in
Progress, July 2014.
[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.
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12.2. Informative References
[I.D-draft-jones-avtcore-private-media-reqts]
Jones, P. et al., "Requirements for Private Media in a
Switched Conferencing Environment", Work in Progress,
March 2015.
[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.
13. Acknowledgments
The authors would like to thank Christian Oien for invaluable input
on this document.
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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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