One document matched: draft-wing-sipping-srtp-key-01.txt
Differences from draft-wing-sipping-srtp-key-00.txt
Network Working Group D. Wing
Internet-Draft Cisco Systems
Intended status: Standards Track F. Audet
Expires: January 9, 2008 Nortel
S. Fries
Siemens AG
H. Tschofenig
Nokia Siemens Networks
July 8, 2007
Disclosing Secure RTP (SRTP) Session Keys with a SIP Event Package
draft-wing-sipping-srtp-key-01
Status of this Memo
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Copyright Notice
Copyright (C) The IETF Trust (2007).
Abstract
Many Secure RTP (SRTP) key exchange mechanisms do not disclose the
SRTP session keys to intermediate SIP proxies. Howevr, these key
exchange mechanisms cannot be used In environments where transcoding,
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monitoring, or call recording are needed. This document specifies a
secure mechanism for a cooperating endpoint to disclose its SRTP
master keys to an authorized party.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Operation . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Learning Name and Certificate of ESC . . . . . . . . . . . 5
3.2. Authorization of ESC . . . . . . . . . . . . . . . . . . . 5
3.3. Sending SRTP Session Keys to ESC . . . . . . . . . . . . . 5
3.4. Scenarios and Call Flows . . . . . . . . . . . . . . . . . 7
4. Grammar . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
5. Security Considerations . . . . . . . . . . . . . . . . . . . 9
5.1. Incorrect ESC . . . . . . . . . . . . . . . . . . . . . . 9
5.2. Risks of Sharing SRTP Session Key . . . . . . . . . . . . 9
5.3. Disclosure Flag . . . . . . . . . . . . . . . . . . . . . 9
5.4. Integrity and encryption of keying information . . . . . . 9
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
8.1. Normative References . . . . . . . . . . . . . . . . . . . 11
8.2. Informational References . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13
Intellectual Property and Copyright Statements . . . . . . . . . . 14
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1. Introduction
This document addresses 2 difficulties with End-to-end encryption of
RTP (SRTP [RFC3711]): transcoding and media recording. When peering
with other networks, different codecs are sometimes necessary
(transcoding a surround-sound codec for transmission over a highly-
compressed bandwidth-constrained network, for example). In some
environments (e.g., stock brokerages and banks) regulations and
business needs require recording calls with coworkers or with
customers. In many environments, quality problems such as echo can
only be diagnosed by listening to the call (analyzing SRTP headers is
not sufficient).
With an RTP stream, transcoding is accomplished by modifying SDP to
offer a different codec through a transcoding device [RFC4117], and
call recording or monitoring can be accomplished with an Ethernet
sniffer listening for SIP and its associated RTP, with a media relay,
or with a Session Border Controller. However, when media is
encrypted end-to-end[I-D.wing-rtpsec-keying-eval], these existing
techniques fail because they are unable to decrypt the media packets.
When a media session is encrypted with SRTP, there are three
techniques to decrypt the media for monitoring or call recording:
1. the endpoint establishes a separate media stream to the recording
device, with a separate SRTP key, and sends the (mixed) media to
the recording device. The disadvantages of this technique
include doubling bandwidth requirements, loss of media recording
facility doesn't cause loss of call (as is required in some
environments). A significant advantage of this technique,
however, is that it's secure: a malicious media recording device
cannot inject media to the connected party on behalf of the
endpoint. Depending on the application requirements it may be
necessary to establish a reliable connection to the recording
device to cope with possible packet loss on the unreliable link,
typically used for media transport.
2. the endpoint relays media through a device which forks a separate
media stream to the recording device. This technique is often
employed by Session Border Controllers, and could also be
employed by TURN servers.
3. Network sniffing devices are used to listen to the SRTP traffic
and correlate SRTP with SIP (with cooperation of call signaling
devices, if the call signaling is encrypted).
This document describes cases (2) and (3) where a cooperating
endpoints publishes its SRTP master keys to an authorized party using
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the SIP Event State Publication Extension [RFC3903]. The mechanism
described in this paper allows secure disclosure of SRTP session keys
to authorized parties so that an endpoints media stream can be
transcoded or decrypted, as needed by that environment.
2. Terminology
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 [RFC2119].
The following terminology is taken directly from SIP Event State
Publication Extension [RFC3903]:
Event Publication Agent (EPA): The User Agent Client (UAC) that
issues PUBLISH requests to publish event state.
Event State Compositor (ESC): The User Agent Server (UAS) that
processes PUBLISH requests, and is responsible for compositing
event state into a complete, composite event state of a resource.
Publication: The act of an EPA sending a PUBLISH request to an ESC
to publish event state.
3. Operation
For transcoding, RTP packets must be sent from and received by a
device which performs the transcoding. When the media is encrypted,
this device must be capable of decrypting the media, performing the
transcoding function, and re-encrypting the media.
ISSUE-1: should we consider providing some or all of the SIP
headers, as well? Some recording functions will need to know the
identity of the remote party. This information could be gleaned
from the SIP proxies, though, and starts to fall outside the
intended scope of this document.
ISSUE-2: The authors have been considering use of MIKEY
[RFC3830], but MIKEY may not be used off the shelf. Certain
changes to the state machine may have to be made (RFC3830
describes the TGK transport rather than SRTP master key
transport).
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3.1. Learning Name and Certificate of ESC
The endpoint will be configured with the AOR of its ESC (e.g.,
"transcoder@example.com"). If S/MIME is used to send the SRTP master
key to the ESC, the endpoint is additionally configured with the
certificate of its ESC.
This configuration is outside the scope of this document, but some
examples are CLI, [I-D.ietf-sipping-config-framework], or
[I-D.ietf-sip-certs].
3.2. Authorization of ESC
Depending on the application, authorization of the key disclosure and
distribution to the ESC may be necessary besides the pure transport
security of the key distribution itself. This may be the case when
the config framework [I-D.ietf-sipping-config-framework] is not
applied and thus the information about the ESC is not known to the
client.
This can be done by providing a SAML extension, according to
[I-D.ietf-sip-saml] in the header of the SUBSCRIBE message. The SAML
assertion shall at least contain the information about the ESC, call
related information to associate the call with the assertion (editors
note: we may also define wildcards here to allow for recordings of
all phone calls for a day, independent of the call) and a reference
to the certificate for the ESC. The latter information is needed to
transport the SRTP Session Key to the ESC in a protected manner, as
described in the section below.
The signature of the SAML assertion should be produced using the
private key of the domain certificate. This certificate MUST have a
SubjAltName which matches the domain of user agent's SIP proxy (that
is, if the SIP proxy is sip.example.com, the SubjAltName of the
domain certificate signing this SAML assertion MUST also be
example.com). Here, the main focus is placed on communication of
clients with the ESC, which belongs to the client's home domain.
3.3. Sending SRTP Session Keys to ESC
SDP is used to describe the media session to the ESC. However, the
existing Security Descriptions [RFC4568] only describes the master
key and parameters of the SRTP packets being sent -- it does not
describe the master key (and parameters) of the SRTP being received,
or the SSRC being transmitted. For transcoding and media recording,
both the sending key and receiving key are needed and in some cases
the SSRC is needed.
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Thus, we hereby extend the existing crypto attribute to indicate the
SSRC. We also create a new SDP attribute, "rcrypto", which is
identical to the existing "crypto" attribute, except that it
describes the receiving keys and their SSRCs. For example:
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32
SSRC=1899
a=rcrypto:1 AES_CM_128_HMAC_SHA1_32
inline:AmO4q1OVAHNiYRj6HmS3JFWNCFqSpTqHWKKIN1Mw|2^20|1:32
SSRC=3289
a=rcrypto:1 AES_CM_128_HMAC_SHA1_32
inline:Hw3JFWNCFqSpTqNiYRj6HmSWKMHAmO4q1KIN1OVA|2^20|1:32
SSRC=4893
Figure 1: Example SDP
The full SDP, including the keying information, is then sent to the
ESC. The keying information MUST be encrypted and integrity
protected. Existing mechanisms such as S/MIME [RFC3261] and SIPS
[I-D.ietf-sip-sips] MAY be used to achieve this goal, or other
mechanisms may be defined.
[[ ISSUE-3: if a endpoint is receiving multiple incoming streams
from multiple endpoints, it will have negotiated different keys
with each of them, and all of that traffic is coming to the same
transport address on the endpoint. Thus, we need a way to
describe the different keys we're using to/from different
transport addresses. One solution is to indicate the remote
transport address. Indicating the remote SSRC is insufficient for
this task, as several SRTP keying mechanisms do not include SSRC
in their signaling (DTLS-SRTP, ZRTP, Security Descriptions).
For example, if there were two remote peers with different keys,
we could signal it like this:
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a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32
192.0.2.1:5678 SSRC=1899 SSRC=3892
a=rcrypto:1 AES_CM_128_HMAC_SHA1_32
inline:AmO4q1OVAHNiYRj6HmS3JFWNCFqSpTqHWKKIN1Mw|2^20|1:32
192.0.2.1:5678 SSRC=3289 SSRC=2813
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:GdUJShpX1ZLEw6UzF3WSJjNzB4d1BINUAv+PSdFc|2^20|1:32
192.0.2.222:2893
a=rcrypto:1 AES_CM_128_HMAC_SHA1_32
inline:6UzF3IN1ZLEwAv+PSdFcWUGdUJShpXSJjNzB4d1B|2^20|1:32
192.0.2.222:2893
Figure 2: Strawman solution
]]
3.4. Scenarios and Call Flows
The following scenarios and call flows depict the assumptions for the
provision of media key disclosure. Figure 3 shows the general setup
within the home domain of the client. Note that the authors assume
that the client only discloses media keys only to an entity in the
client's home network rather than to an arbitrary entity in the
visited network.
+--------+ +-------+ +-------+ +--------+
| Client | | Proxy | | ESC | | MedRec |
+--------+ +-------+ +-------+ +--------+
| | | |
+---------------+--------------+--------------+
Figure 3: Network Topology
Based on this setup there are different options to realize the key
disclosure, depending on the environment. In the following two
approaches are distingusihed.
Publishing media keys to the ESC
This requires that the configuration management provides the ESC
configuration data (e.g., certificate, policy) in a secure way to
the client. As stated above, this configuration is outside the
scope of this document, but an example can be found in
[I-D.ietf-sipping-config-framework]. The key disclosure in this
approach uses the PUBLISH method to disclose the key to the ESC
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according to a given policy.
+--------+ +-------+ +-------+ +--------+
| Client | | Proxy | | ESC | | MedRec |
+--------+ +-------+ +-------+ +--------+
| | | |
| - REGISTER -> | | |
| <- 200 OK -- | | |
| | | |
| -- INVITE --> | | |
| <- 200 OK -- | | |
| | | |
| -- PUBLISH (Key Data) --> | |
| | | |
Using SAML assertions for ESC contact
In this approach authorization is provided via a SAML assertion,
see [I-D.ietf-sip-saml], indicating which ESC is allowed to
perform call recording of a single or a set of calls, depending on
the content of the assertion. Here a SAML assertion is provided
as part of the SUBSCRIBE message, send from the ESC to the client.
The assertion needs to provide at least the call relation, or a
time intervall for which media recoding is going to be performed.
The SAML assertion is signed with the private key associated with
the domain certificate, which is in possession of the
authentication service. The call flow would look like following:
+--------+ +-------+ +-------+ +--------+
| Client | | Proxy | | ESC | | MedRec |
+--------+ +-------+ +-------+ +--------+
| | | |
| - REGISTER -> | | |
| <- 200 OK -- | | |
| | | |
| <- SUBSCIBE (call forming) - | |
| + SAML assertion | |
| | | |
| -- INVITE --> | | |
| <- 200 OK -- | | |
| | | |
| -- NOTIFY (Key Data) --> | |
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4. Grammar
[[Grammar will be provided in a subsequent version of this
document.]]
5. Security Considerations
5.1. Incorrect ESC
Insertion of the incorrect public key of the SRTP ESC will result in
disclosure of the SRTP session key to an unauthorized party. Thus,
the UA's configuration MUST be protected to prevent such
misconfiguration. If the configuration or the ESC's certificate are
obtained over the network [I-D.ietf-sipping-config-framework],
[I-D.ietf-sip-certs], the certificate MUST be suitably authenticated
and integrity protected.
5.2. Risks of Sharing SRTP Session Key
A party authorized to obtain the SRTP session key can listen to the
media stream and could inject data into the media stream as if it
were either party. The alternatives are worse: disclose the
device's private key to the transcoder or media recording device, or
abandon using secure SRTP key exchange in environments that require
media transcoding or media recording. As we wish to promote the use
of secure SRTP key exchange mechanisms, disclosure of the SRTP
session key appears the least of these evils.
5.3. Disclosure Flag
Secure SRTP key exchange techniques which implement this
specification SHOULD provide a "disclosure flag", similar to that
first proposed in Appendix B of [I-D.zimmermann-avt-zrtp].
5.4. Integrity and encryption of keying information
The mechanism describe in this specification relies on protecting and
encrypting the keying infomation. There are well known mechanism to
achieve that goal.
Using SIPS to convey the SRTP key exposes the SRTP master key to all
SIP proxies between the Event Publication Agent (ESC, the SIP User
Agent) and the Event State Compositor (ESC). S/MIME allows
disclosing the SRTP master key to only the ESC.
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6. IANA Considerations
New SSRC extension of the "crypto" attribute, and the new "rcrypto"
attribute will be registered here.
7. Examples
This is an example showing a SIPS AOR for the ESC. This relies on
the SIP network providing TLS encryption of the SRTP master keys to
the ESC.
PUBLISH sips:recorder@example.com SIP/2.0
Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
To: <sips:recorder@example.com>
From: <sip:dan@example.com>;tag=1234wxyz
Call-ID: 81818181@pua.example.com
CSeq: 1 PUBLISH
Max-Forwards: 70
Expires: 3600
Event: srtp
Content-Type: application/sdp
Content-Length: ...
v=0
o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com
s=-
c=IN IP4 192.0.2.101
t=0 0
m=audio 49172 RTP/SAVP 0
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32
a=rcrypto:1 AES_CM_128_HMAC_SHA1_32
inline:AmO4q1OVAHNiYRj6HmS3JFWNCFqSpTqHWKI8K1Mw|2^20|1:32
a=rtpmap:0 PCMU/8000
Figure 6: Example with "SIPS:" AOR
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This is an example showing an S/MIME-encrypted transmission to the
media recorder's AOR, recorder@example.com. The data enclosed in "*"
is encrypted with recorder@example.com's public key.
PUBLISH sip:recorder@example.com SIP/2.0
Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
To: <sip:recorder@example.com>
From: <sip:dan@example.com>;tag=1234wxyz
Call-ID: 81818181@pua.example.com
CSeq: 1 PUBLISH
Max-Forwards: 70
Expires: 3600
Event: srtp
Content-Type: application/pkcs7-mime;smime-type=enveloped-data;
name=smime.p7m
Content-Transfer-Encoding: binary
Content-ID: 1234@atlanta.example.com
Content-Disposition: attachment;filename=smime.p7m;handling=required
Content-Length: ...
******************************************************************
* (encryptedContentInfo) *
* Content-Type: application/sdp *
* Content-Length: ... *
* *
* v=0 *
* o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com*
* s=- *
* c=IN IP4 192.0.2.101 *
* t=0 0 *
* m=audio 49172 RTP/SAVP 0 *
* a=crypto:1 AES_CM_128_HMAC_SHA1_32 *
* inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 *
* a=rcrypto:1 AES_CM_128_HMAC_SHA1_32 *
* inline:AmO4q1OVAHNiYRj6HmS3JFWNCFqSpTqHWKI8K1Mw|2^20|1:32 *
* a=rtpmap:0 PCMU/8000 *
* *
******************************************************************
Figure 7: Example with S/MIME-encrypted SDP
8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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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.
[RFC3903] Niemi, A., "Session Initiation Protocol (SIP) Extension
for Event State Publication", RFC 3903, October 2004.
[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.
8.2. Informational References
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[I-D.wing-rtpsec-keying-eval]
Audet, F. and D. Wing, "Evaluation of SRTP Keying with
SIP", draft-wing-rtpsec-keying-eval-02 (work in progress),
February 2007.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[I-D.ietf-sipping-config-framework]
Petrie, D. and S. Channabasappa, "A Framework for Session
Initiation Protocol User Agent Profile Delivery",
draft-ietf-sipping-config-framework-12 (work in progress),
June 2007.
[I-D.ietf-sip-sips]
Audet, F., "The use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", draft-ietf-sip-sips-05 (work
in progress), June 2007.
[I-D.ietf-sip-certs]
Jennings, C., "Certificate Management Service for The
Session Initiation Protocol (SIP)",
draft-ietf-sip-certs-03 (work in progress), March 2007.
[I-D.ietf-sip-saml]
Tschofenig, H., "SIP SAML Profile and Binding",
draft-ietf-sip-saml-02 (work in progress), May 2007.
[I-D.zimmermann-avt-zrtp]
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Zimmermann, P., "ZRTP: Media Path Key Agreement for Secure
RTP", draft-zimmermann-avt-zrtp-03 (work in progress),
March 2007.
[RFC4117] Camarillo, G., Burger, E., Schulzrinne, H., and A. van
Wijk, "Transcoding Services Invocation in the Session
Initiation Protocol (SIP) Using Third Party Call Control
(3pcc)", RFC 4117, June 2005.
Authors' Addresses
Dan Wing
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134
USA
Email: dwing@cisco.com
Francois Audet
Nortel
4655 Great America Parkway
Santa Clara, CA 95054
USA
Email: audet@nortel.com
Steffen Fries
Siemens AG
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: steffen.fries@siemens.com
Hannes Tschofenig
Nokia Siemens Networks
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: Hannes.Tschofenig@nsn.com
URI: http://www.tschofenig.com
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