One document matched: draft-dcsgroup-sip-arch-00.txt
SIP Working Group W. Marshall
Internet Draft K. Ramakrishnan
Document: <draft-dcsgroup-sip-arch-00.txt> AT&T
Category: Informational
E. Miller
G. Russell
CableLabs
B. Beser
M. Mannette
K. Steinbrenner
3Com
D. Oran
Cisco
J. Pickens
Com21
P. Lalwaney
J. Fellows
General Instrument
D. Evans
Lucent Cable
K. Kelly
NetSpeak
F. Andreasen
Telcordia
October, 1999
Architectural Considerations for Providing Carrier Class Telephony
Services Utilizing SIP-based Distributed Call Control Mechanisms
Status of this Memo
This document is an Internet-Draft and is NOT offered in accordance
with Section 10 of RFC2026[1], and the author does not provide the
IETF with any rights other than to publish as an Internet-Draft.
Internet-Drafts are working documents of the Internet Engineering
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progress."
DCS Group Category: Informational - Expiration 4/30/00 1
DCS Architecture October 1999
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The distribution of this memo is unlimited. It is filed as <draft-
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comments to the authors.
1. Abstract
This document provides an overview of a SIP-based Distributed Call
Signaling (DCS) architecture to support carrier class packet-based
voice, video, and other real time multimedia services. Companion
documents [3,4,5,6,7] address a specific set of SIP 2.0 protocol
extensions and usage rules that are necessary to implement the DCS
architecture in an interoperable fashion.
The DCS architecture takes advantage of endpoint intelligence in
supporting telephony services without sacrificing the network's
ability to provide value through mechanisms such as resource
management, lookup of directory information and translation
databases, routing services, security protection, and privacy
enforcement. At the same time, the architecture provides
flexibility to allow evolution in the services that may be provided
by endpoints and the network.
DCS also takes into account the need to manage access to network
resources and account for resource usage. The SIP usage rules
defined in the accompanying IDs specifically address the
coordination between Distributed Call Signaling and dynamic quality
of service control mechanisms for managing resources over the access
network. In addition, the DCS architecture defines the interaction
needed between network provided call controllers, known as a "DCS-
proxy" (also called Gate Controller in this draft on occasion) for
supporting these services.
2. Conventions used in this document
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 [2].
3. Introduction
This document provides an overview of a SIP-based Distributed Call
Signaling (DCS) architecture to support carrier class packet-based
voice, video, and other real time multimedia services. The DCS
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DCS Architecture October 1999
architecture and the corresponding SIP protocol enhancements
(described in companion documents) are being developed as part of
the cable industry's PacketCable initiative, managed out of
CableLabs (see www.cablelabs.com). PacketCable is defining a series
of interface specifications that will enable vendors to develop
interoperable products for providing internet telephony and other
multimedia services over DOCSIS-enabled cable data networks.
The DCS architecture described herein has its roots in the DOSA work
performed by AT&T Laboratories ["Distributed Open Signaling
Architecture"; Kalmanek, Marshall, Mishra, Nortz, Ramakrishnan, et
al.; October, 1998]. AT&T contributed DOSA to PacketCable in March,
1999 as a framework proposal for DCS. Earlier exploratory work by
AT&T and a select group of vendors had demonstrated that the DOSA
concepts could be expressed in terms of SIP signaling messages.
Since the AT&T submission, a relatively large group of vendors have
cooperated in an intensive effort to develop the DCS architecture
and SIP protocol extensions described here and in the accompanying
protocol drafts. Although DCS was originally designed with cable
access networks in mind, the SIP signaling enhancements have general
applicability to carrier class VOIP services running over QoS
enabled IP networks.
The authors are submitting this draft to the IETF in order to
provide general information regarding the DCS architecture and to
convey the motivation behind the SIP enhancements recommended in the
accompanying protocol drafts. We believe that incorporation of the
concepts and mechanisms described in this set of drafts by the IETF
into the SIP standard will significantly enhance SIP's ability to
function as a carrier-class signaling protocol. Such an enhancement
to SIP would undoubtedly aid in its widespread acceptance and
deployment. The PacketCable Draft Specification for DCS is available
from the CableLabs website at:
ftp://ftp.cablelabs.com/pub/ietfdocs/dcsdraft.pdf.
3.1 Background and Motivation
The design of the Distributed Call Signaling (DCS) architecture
recognizes the trend towards use of packet networks as the
underlying framework for communications. These networks will
provide a broad range of services, including traditional best-effort
data service as well as enhanced, value-added services, such as
telephony. At the same time, improvements in silicon will reinforce
the trend towards increased functionality in endpoints. These
intelligent endpoints will take advantage of the widespread
availability of packet networks to enable a rich set of applications
and services for users.
However, when the network is used for real-time telephony
applications, it is essential to have service differentiation at the
IP layer. The ability to control and monitor usage is needed for
the provider to be able to provide service differentiation and to
derive revenue from the enhanced services. At the same time, the
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DCS Architecture October 1999
availability of best effort communications and the migration of
functionality to the endpoints pose a challenge to the provider to
find incentives for users to use or pay for enhanced services.
We see three key functions that a provider can offer, as incentives
to use enhanced services. First, the network service provider has
the unique ability to manage and provide network layer quality of
service. When users depend on the quality of the service, as with
telephony, there is a strong incentive to use the enhanced service,
rather than a best effort service. Second, the network service
provider can play an important role as a trusted intermediary. This
includes ensuring the integrity of call routing, as well as ensuring
both the accuracy and the privacy of information that is exchanged.
The service provider can also add value by ensuring that services
are provided consistently and reliably, even when an endpoint is
unavailable. Finally, there are a number of services that may be
offered more efficiently by the network service provider rather than
in endpoints. For example, conference bridging may be more cost
effective to implement in a multi-point bridge rather than in every
endpoint attached to the network.
A key contribution of the DCS architecture is a recognition of the
need for coordination between call signaling, which controls access
to telephony specific services, and resource management, which
controls access to network-layer resources. This coordination is
designed to meet the user expectations and human factors associated
with telephony. For example, the called party should not be
alerted until the resources necessary to complete the call are
available. If resources were not available when the called party
picked up, the user would experience a call defect. In addition,
users expect to be charged for service only after the called party
answers the phone. As a result, usage accounting starts only after
the called party picks up. Coordination between call signaling and
resource management is also needed to prevent fraud and theft of
service. The coordination between DCS and Dynamic QoS protocols
ensures that users are authenticated and authorized before receiving
access to the enhanced QoS associated with the telephony service.
It is important to be able to deploy a residential telephone service
at very large scale, cost-effectively. To achieve this, DCS
minimizes the messaging overhead on network call servers, and does
not require these servers to maintain call state for active calls.
Once a call is established, call state is maintained only where it
is needed: at the endpoints that are involved in the call, and at
the ERs in the bearer path that are providing differentiated service
to the media flow. This allows the network call servers to scale to
support more users, and imposes less stringent reliability
requirements on those servers.
DCS is also designed so that calling users receive consistent
service even when a called endpoint is unavailable. For example,
when an endpoint is unavailable service logic in a network call
server can forward telephone calls to a voice mailbox.
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3.2 Requirements And Design Principles
In this section, we briefly describe the application requirements
that led to a set of DCS signaling design principles. In its most
basic implementation, DCS supports a residential telephone service
comparable to the local telephone services offered today. In
addition to the commonly used service features that need to be
supported, there are important requirements in the areas of
reliability, performance, and scalability that influence the
signaling architecture. Supporting an IP telephony service
comparable to the telephony service offered today requires enhanced
bearer channel and signaling performance, including:
@ Low delay - end-to-end packet delay must be small enough that it
does not interfere with normal voice conversations. The ITU
recommends no greater than 300 ms roundtrip delay for telephony
service.
@ Low packet loss - packet loss must be small enough to not
perceptibly impede voice quality or performance of fax and voice
band modems.
@ Short post-dial delay - the delay between the user dialing the
last digit and receiving positive confirmation from the network
must be short enough that users do not perceive a difference with
post-dial delay in the circuit switched network or believe that
the network has failed.
@ Short post pickup delay - the delay between a user picking up a
ringing phone and the voice path being cut through must be short
enough so that the "hello" is not clipped.
We identify a number of key design principles that arise from the
requirements and philosophy outlined above.
1. Providing differentiated network-layer quality of service is
essential, while allowing the provider to derive revenues from the
use of such differentiated services.
2. The architecture should allow, and even encourage, implementation
of services and features in the intelligent endpoints, where
economically feasible, while still retaining value in the network
and network-based services
3. The architecture must ensure that the network is protected from
fraud and theft of service. The service provider must be able to
authenticate users requesting service and ensure that only those
authorized to receive a particular service be able to obtain it.
4. The architecture must enable the service provider to add value by
supporting the functions of a trusted intermediary. This includes
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DCS Architecture October 1999
protecting the privacy of calling and called party information,
and ensuring the accuracy of the information that is provided in
messages from the network.
5. The architecture must enable the service provider to give a
consistent view of basic services and features even when customer
premise equipment is unavailable, and allow users to take
advantage of functionality that is provided in the network, when
it is cost-effective and easy to use.
6. The architecture must be implementable, cost-effectively, at very
large scale.
3.3 Distributed Call Signaling Architecture
The Distributed Call Signaling Architecture follows the principles
outlined above to support a robust telephony service. Figure 1
introduces the key components in the architecture.
The architecture assumes a broad range of DCS-compliant endpoints
that provide telephony service to the user including Media Terminal
Adapters (MTAs) that may be integrated with a Cable Modem or is a
standalone device, as well as other endpoints such as personal
computers. The access network interfaces to an IP backbone through
a system we refer to as the Edge Router (ER). The ER is the first
trusted element within the provider's network and is considered to
be the edge of the network for providing access to differentiated
quality of service. The ER performs resource management, acts as a
policy enforcement point and as a source of billing information.
DCS-proxies (GCs) process call signaling messages and support number
translation, call routing, feature support and admission control.
In the context of SIP, a DCS-proxy is a SIP proxy that is involved
in processing and forwarding of SIP requests. GCs act as trusted
decision points for controlling when resources are committed to
particular users. Media servers represent network-based components
that operate on media flows to support the service. Media servers
perform audio bridging, play terminating announcements, provide
interactive voice response services, etc. Finally, PSTN gateways
interface to the Public Switched Telephone Network.
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DCS Architecture October 1999
+-----+
| MTA | MTA: media terminal adapter
+-----+
| ER: Edge Router
Access Network |
|
+----+
| ER |
+----+
| +-----------+
|----| DCS Proxy |
| +-----------+
|
| +------------+
Backbone Network |----|Media Server|
| +------------+
|
| +------------+
|----|PSTN Gateway|
| +------------+
+----+
| ER |
+----+
|
Access Network |
|
+-----+
| MTA |
+-----+
Figure 1: A Simple view of Components of an IP Telephony
Architecture used in a HFC Cable Environment.
Telephony endpoints are considered to be "clients" of the telephony
service. Consistent with the design principles, the architecture
allows a range of services to be implemented by intelligent
endpoints. They collect dialed digits, participate in signaling and
contain the service logic required for basic call setup and feature
support. Endpoints also participate in end-to-end capability
negotiation. However, endpoints are not trusted to provide accurate
information to the network or to keep information that is received
private, except when it is in the endpoint's best interests to do
so.
Access to network resources on a differentiated basis is likely to
be controlled by the service provider. The ER receives resource
management requests from endpoints, and is responsible for ensuring
that packets are provided the QoS they are authorized to receive
(either through packet marking, or through routing and queueing the
packets as a specific QoS assured flow). The ER requires
authorization from a network entity (on a call-by-call basis for the
telephony service) before providing access to enhanced QoS for an
end-to-end IP flow. The obvious point where this policy and control
function resides is the DCS-proxy (also called gate-controller,
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DCS Architecture October 1999
because of this responsibility for managing access to enhanced QoS).
Thus, the ER is able to ensure that enhanced QoS is only provided
for end-to-end flows that have been authorized and for which usage
accounting is being done. Since the ER knows about the resource
usage associated with individual IP flows, it generates the usage
events that allow a user to be charged for service.
We introduce the concept of a "gate" in the ER, which manages access
to enhanced quality of service. The gate is a packet classifier and
policer that ensures that only those IP flows that have been
authorized by the DCS-proxy are granted access to enhanced QoS in
the access and backbone networks. Gates are "opened" selectively
for a flow. For the telephony service, they are opened for
individual calls. Opening a gate involves an admission control
check that is performed when a resource management request is
received from the endpoint for an individual call, and it may
involve resource reservation in the network for the call if
necessary. The packet filter in the gate allows a flow of packets to
receive enhanced QoS for a call from a specific IP source address
and port number to a specific IP destination address and port
number.
The DCS-proxy, in addition to implementing many of the call control
functions, is responsible for the policy decision regarding whether
the gate should be opened. DCS sets up a gate in advance of a
resource management message. This allows the policy function, which
is at the DCS-proxy, to be "stateless" in that it does not need to
know the state of calls that are already in progress.
DCS-proxies are typically organized in domains. A DCS-proxy is
responsible for a set of endpoints and the associated ERs. While
endpoints are not trusted, there is a trust relationship between the
ER and its associated DCS-proxy, since the DCS-proxy plays a role as
a policy server controlling when the ER can provide enhanced QoS
service. There is also a trust relationship among DCS-proxies.
Details of the security model and mechanisms are work in progress.
The DCS-proxy is designed as a simple transaction server, so that
failure of a DCS-proxy does not affect calls in progress. A domain
will likely have both a primary and a secondary DCS-proxy. If the
primary DCS-proxy fails, only calls in a transient state are
affected. The endpoints involved in those calls will time out and
retry. All active calls are unaffected. This is possible because
the DCS-proxy retains no call state for stable calls. We believe
this design makes the DCS-proxy efficient and highly scalable, and
keeps the reliability requirements manageable.
DCS supports inter-working with the circuit switched telephone
network through PSTN gateways. A PSTN gateway may be realized as a
combination of a media controller, media gateway, and a signaling
gateway. A media gateway acts as the IP peer of an endpoint for
media packets, converting between the data format used over the IP
network and the PCM format required for transmission over the PSTN.
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The signaling gateway acts as the IP peer of an endpoint for
signaling packets, providing signaling inter-working between DCS and
conventional telephony signaling protocols such as ISUP/SS7. A media
gateway control protocol is used to control the operation of the
media gateway from the signaling gateway.
There are additional system elements that may be involved in
providing the telephony service. For example, the DCS-proxy may
interface with other servers that implement the authorization or
translation functions. Similarly, three way calling may be
supported using media servers in the network.
3.4 Basic Call Flow
Figure 2 presents a high-level overview of a basic MTA-to-MTA call
flow in DCS. Each MTA is associated with a DCS-proxy, which acts as
a SIP proxy. When a user goes off-hook and dials a telephone
number, the originating MTA (MTA-o) collects the dialed digits and
sends the initial call message, called an INVITE in SIP, to the
"originating" DCS-proxy (DP-o). DP-o verifies that MTA-o is a valid
subscriber of the telephony service (using authentication
information in the INVITE message) and determines whether this
subscriber is authorized to place this call. DP-o then translates
the dialed number into the address of a "terminating" DCS-proxy (DP-
t) and forwards the INVITE message to it.
We assume that the originating and terminating DCS-proxies trust
each other. DP-o augments the INVITE message that it forwards with
additional information, such as billing information containing the
account number of the caller. DP-t then translates the dialed
number into the address of the terminating MTA (MTA-t) and forwards
the INVITE message to MTA to notify it about the incoming call.
The initial INVITE message invokes call feature handling at the
destination MTA, such as call-forwarding. Assuming that the call is
not forwarded, MTA-t negotiates the coding style and bandwidth
requirements for the media streams. The 200 OK response to the
initial INVITE is forwarded back through the DCS-proxies.
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MTA-o ER-o DP-o DP-t ER-t MTA-t
| Invite | | | | |
|(Stage1) | | Invite | | |
|----------|--------->|(Stage1) | | |
| | |--------->| | |
| | | | Invite | |
| | | |(Stage1) | |
| | | |----------|--------->|
| | | | | 200 OK |
| | | |<---------|----------|
| | | | | |
| | Gate | 200 OK |Gate Setup| |
| | Setup |<---------|--------->| |
| |<---------| | | |
| 200 OK | | | | |
|<---------|----------| | | |
| ACK | | | | |
|----------|----------|----------|----------|--------->|
| | | Reserve | | |
| | | Resources| | |
|<---------|----------|----------|----------|--------->|
| Invite | | | | |
|----------|--------- |----------|----------|--------->|
| | | | | 180 Ring |
|<---------|----------|----------|----------|----------|
| | | | | | User
| | | | | 200 OK | Answers
|<---------|----------|----------|----------|----------|
| ACK | | | | |
|----------|----------|----------|----------|--------->|
| | | Active | | |
| | | Call | | |
|<---------|--------- |----------|----------|--------->|
| | | | | | User
| | | | | BYE | Hangs Up
|<---------|----------|----------|----------|----------|
| | | | | 200 OK |
|----------|----------|----------|----------|--------->|
Figure 2: A Basic Call Flow, including Resource Management functions
In the figure, MTA-t sends a 200 OK message to DP-t. The 200 OK
contains a subset of the capabilities in the INVITE message that are
acceptable to MTA-t. DP-t sends a GATE-SETUP message to the
terminating ER (ER-t), conveying policy instructions allowing ER-t
to open a gate for the IP flow associated with this phone call. The
GATE_SETUP message contains billing information containing the
account number of the subscriber that will pay for the call.
DP-t forwards the 200 OK to DP-o. DP-o sends a GATE-SETUP message
to the originating ER (ER-o) to indicate that it can open a gate for
the IP flow associated with the phone call. Finally, DP-o forwards
200 OK to MTA-o. The initial INVITE request and 200 OK response
contain a SIP Contact header to indicate the IP address of the
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DCS Architecture October 1999
remote MTA to be used for subsequent end-to-end SIP signaling
exchanges. MTA-o acknowledges the 200 OK directly to MTA-t.
Once the initial INVITE/200 OK exchange has completed, both MTAs
reserve the resources that will be needed for the media streams.
Once MTA-o has successfully made its reservation, it sends a second
INVITE message to MTA-t containing a command to ring the destination
telephone. If MTA-t successfully reserved the resources needed for
the call, it responds with a 180 Ringing to indicate that the phone
is ringing, and that the calling party should be given a ringback
call progress tone. When the called party answers, by going off-
hook, MTA-t sends a 200 OK final response, which MTA-o acknowledges.
At this point the resources that were previously reserved are
committed to this conversation, and the call is "cut through."
Either party can terminate the call. An MTA that detects an on-hook
sends a SIP BYE message to the remote MTA, which is acknowledged.
4. Resource Management
DCS's resource management protocols distinguish between two phases:
Reserve and Commit. During the first phase, resources are reserved
but are not yet made available to the endpoint. This ensures that
resources are available before ringing the far-end telephone. The
second phase, which commits resources, is initiated after ringing
the far end telephone and after the called party picks up. At this
point, resources are committed and made available to the endpoint,
and recording is started so that the user can be billed for usage.
The use of a two-phase protocol is essential because of the unique
requirements associated with human communication, such as telephony.
In addition to ensuring that resources are available before ringing
the phone, it also preserves the semantics of billing that users are
accustomed to, whereby usage recording is not started until the
called party picks up the phone. Backbone resources are reserved
and allocated in the first phase of the two-phase resource
reservation protocol. This is important in order to limit the
impact of backbone resource management on post-pickup delay (this
minimizes the likelihood of clipping the first few syllables of the
conversation).
5. Distributed Call State
In order to provide enhanced services to millions of endpoints, we
need an architecture that can be implemented cost-effectively at
very large scale. Just as we enable flexibility by exploiting
intelligence at the endpoints, services can be provided in a
scaleable manner by storing the state associated with applications
at the endpoints, rather than in network servers. Especially with
telephony, endpoints are directly involved in handling calls and
therefore need to maintain and use call state. In contrast, while
network servers may need to be involved when setting up a call to
gain access to enhanced QoS, there is no fundamental need for those
servers to be involved throughout the lifetime of the call.
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DCS Architecture October 1999
Maintaining state for every call at network servers, while
achievable, increases the reliability requirements and load on the
servers. The less state kept in the network, the better.
As a result, the DCS-proxy in DCS are designed to be stateless
transaction servers. When a DCS-proxy processes a service request
from an endpoint, it maintains state until the transaction is
complete, but does not maintain any per-call state about active
calls in the network. There are two major advantages to this
design. First the reliability of the service does not depend on the
reliability of an individual DCS-proxy. A DCS-proxy can fail without
affecting calls that are currently in progress. Second, it removes
many complex synchronization problems where two (or more) entities
need to have simultaneously accurate information. Since
interactions with the DCS-proxies are simple stateless transactions,
it is not necessary for consecutive calls to be processed by the
same DCS-proxy. DCS-proxy crashes affect only the transient calls
(the number of calls in the process of being set up), and not stable
conversations. Further, it is likely that most calls in a transient
state can be recovered and successfully established through a backup
or spare DCS-proxy using endpoint retransmission, with no explicit
synchronization protocol required between the DCS-proxies. We
believe this design principle will enable us to operate in very
large scale, cost effectively. Furthermore it places the function
of managing the state of a call where it belongs - at the endpoint.
An existing call can only be affected by failures along the path or
by failure of the endpoints: there are no unnecessary elements
involved in a call.
We note that there are many services that involve the use of servers
or proxy endpoints that communicate directly with clients. Since
these endpoints are directly involved in providing service, it is
necessary and appropriate for them to maintain state. Examples of
proxy endpoints include application layer firewalls, caching
servers, transcoders, network-based conference bridges, interactive
voice response systems, and PSTN gateways. The DCS architecture
models these as end-points, that maintain appropriate call state.
We now turn to the mechanisms that allow us to avoid state in the
DCS-proxies. The DCS-proxy stores state information about an
endpoint's calls at that endpoint in an object called a DCS-State
string. DCS-state information are both encrypted and signed by the
DCS-proxy to ensure the privacy and the integrity of the information
contained in the string. When needed, the endpoint provides the
DCS-State to the DCS-proxy, which can use the information to provide
additional functionality. Because the DCS-State contains
information encrypted by the DCS-proxy, the information it contains
is trusted by the network even though the endpoint itself is not
trusted. In addition, the DCS-proxy stores service-specific opaque
data associated with a call at the edge router. Since charging for
telephony services is tightly tied to the use of resources, this
information is best stored at the edge router, where knowledge of
resource usage exists.
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The endpoint returns the information to the DCS-proxy when it is
needed to implement specific features. The endpoint cannot
interpret the information in the string and any attempt to tamper
with it can be detected by the DCS-proxy. There are a number of
examples that arise in the implementation of telephony features.
For example, when a user at an endpoint wants to return the last
call (e.g., by dialing *69 on a traditional telephone) the "call
return" function is invoked. If the user had subscribed to the
caller ID service feature, the terminating endpoint could store the
information (phone number or IP address) associated with the last
call. However, it may be the case that the user does not subscribe
to the feature, or the originator of the previous call may have
requested that this information be blocked in order to retain
privacy. In this case, call return can be implemented, while keeping
the caller's identity private, by using a DCS-state string.
6. DCS Proxy - DCS Proxy Communications
DCS-proxies implement a set of service-specific control functions
required to support the telephony service:
@ Authentication and authorization: Since services are only provided
to authorized subscribers, DCS-proxies authenticate signaling
messages and authorize requests for service on a call-by-call
basis.
@ Name/number translation and call routing: DCS-proxies translate
dialed E.164 numbers, or names, to a terminating IP address based
on call routing logic to support a wide range of call features.
@ Service-specific admission control: DCS-proxies can implement a
broad range of admission control policies for the telephony
service. For example, DCS-proxies may provide precedence for
particular calls (e.g., 911 calls). Admission control may also be
used to implement overload control mechanisms, e.g. to restrict
the number of calls to a particular location or to restrict the
frequency of call setup to avoid signaling overload.
@ Signaling and service feature support: While many service features
are implemented by endpoints, the DCS-proxy also plays a role in
feature support. DCS signaling provides a set of service
primitives to end-points that are mediated by the DCS-proxy. The
DCS-proxy is involved in implementing service features that depend
on the privacy of calling information, e.g., caller-ID blocking.
It also plays a role in supporting service features that require
users to receive a consistent view of feature operation even when
an endpoint is down. For example, while an endpoint may normally
participate in call forwarding, the DCS-proxy can control call
forwarding on behalf of an endpoint when the endpoint is down.
End-points MTA-o and MTA-t communicate through the DCS-Proxies DP-o
and DP-t, as shown in Figure 2. The interface of concern in this
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DCS Architecture October 1999
section is the one between the DCS-Proxies DP-o and DP-t. In
contrast to a true stateless SIP proxy, the DCS-Proxy maintains
transaction state. During the interval that a call is being setup, a
DCS-Proxy keeps state related to a request until a response is
received.
For each call made to a phone number, DP-o may need to perform the
functions needed for Local Number Portability (LNP). If a LNP
database lookup is performed and the resulting dialed string is
modified, DP-o must modify the Request-URI to include the result of
the LNP lookup. When using a two-stage call setup, as described in
[3], the information provided in the INVITE(Stage1), the originating
proxy DP-o generates and stores the DCS-State: header. This
information is intended to be sent to endpoint MTA-o and included
with the final response that is returned to MTA-o. The originating
DCS-Proxy, DP-o, may then use the call state information provided to
it in the DCS-State: header to manipulate call-legs when requested
by MTA-o.
As with conventional SIP proxies, DP-o adds its address to the top
of the Via: header list with a branch=1 field when forwarding the
request (e.g., INVITE(Stage1)). In addition, to support billing
functions for a carrier, DP-o appends opaque pieces of information
called the Billing-Info: and Billing-ID:. In addition, to support
the resource management functions (such as manipulating Gates for
resource management in concert with call-leg manipulation), a Gate-
Location: header is included. This allows for the subsequent
generation of requests for access network QoS by the end-points.
We also depend on originating DCS-Proxy, DP-o to be responsible for
manipulating call legs. For instance, when a call is being
forwarded, information about the new destination that the call is
being forwarded to is provided by DP-t to DP-o. The new INVITE is
then issued from DP-o. The information exchanged between the DCS-
proxies enables such a function to be performed.
7. Privacy
Many conventional telephony systems have the ability to provide
information about the identity of the calling party to the called
party before the latter accepts the call (such a capability is
typically termed "Caller-ID"). Systems that support Caller-ID
usually provide a mechanism that allows the calling party to
instruct the network to refrain from delivering this information to
the destination.
In order for an IP-based network to provide a caller with a similar
capability, a new SIP header is needed to signal the desire for
anonymity to the network elements that would otherwise provide the
caller's identity to the destination party. If a caller desires to
remain anonymous, several additional changes to standard SIP are
necessary.
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DCS Architecture October 1999
In the cable IP network, the triplet {From:, To:, Call-ID:} is used
to identify a call leg. Because call state information is pushed to
the edge of the network, this information must be delivered to the
destination endpoint.
The SIP From: header normally contains information that identifies
the caller. In order to hide the identity of the caller, the From:
header information is encrypted with the originating endpoint's key.
The destination endpoint does not possess the key to decrypt the
From: information. No new syntax for SIP is introduced here.
Normally, the SIP Call-ID: header also contains information about
the caller. In the DCS architecture, to support privacy in cable-
based IP telephony networks the value of Call-ID: header is a
cryptographic hash string that contains no information about the
user.
Since all the normally-available mechanisms for passing information
about the caller are no longer available, a new SIP header is used
(Caller:) to pass the caller's identity to the destination if
privacy has not been requested. This header is introduced that
contains the information that would normally be present in the From:
header; the network passes it to the destination endpoint only if
the caller has not requested anonymity.
In addition to the usual privacy elements provided by telephone
systems, IP-based systems must implement methods of hiding the
source IP address from the destination if the caller requires
privacy. The entire address must be obscured, since even a few
address bits may provide partial location information. Likewise, IP
addresses of the destination should not be revealed to the caller,
in order to maintain privacy of transfer destinations.
IP addresses typically appear in the Contact: header; they also
appear in SDP descriptions contained in SIP messages. These must all
be protected. The mechanism we choose is an application-level
anonymizer that inspects the SIP call signaling messages and
replaces any identifying information contained therein in a
consistent manner and in such a way that when the messages are
delivered to the destination endpoint any identifying information
has been replaced with fields that obscure the identity of the party
seeking privacy.
This mechanism does not require any modification to the call
signaling initiated by the endpoints: the application-level
anonymizer performs these functions silently within the network.
8. Security Considerations
Detailed security considerations related to this architecture will
be addressed in a future companion draft.
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9. References
1. Bradner, S., "The Internet Standards Process -- Revision 3", BCP
9, RFC 2026, October 1996.
2. Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
3. "Integration of Resource Management and Call Signaling for IP
Telephony", Internet Draft: <draft-dcsgroup-sip-resource-00.txt>,
June 1999.
4. "SIP Extensions for Caller Privacy", Internet Draft: <draft-
dcsgroup-sip-privacy-00.txt>, June 1999.
5. "SIP proxy-to-proxy extensions for supporting Distributed Call
State", Internet Draft: <draft-dcsgroup-sip-proxy-proxy-00.txt>,
June 1999.
6. "SIP extensions for supporting Distributed Call State", Internet
Draft: <draft-dcsgroup-sip-state-00.txt>, June 1999.
7. "Integration of Call Authorization and Call Signaling for IP
Telephony", Internet Draft: <draft-dcsgroup-sip-call-auth-
00.txt>, June 1999.
10. Acknowledgments
The Distributed Call Signaling work in the PacketCable project is
the work of a large number of people, representing many different
companies. The authors would like to recognize and thank the
following for their assistance: John Wheeler, Motorola; David
Boardman, Daniel Paul, Arris Interactive; Bill Blum, Jon Fellows,
Jay Strater, Jeff Ollis, Clive Holborow, General Instruments; Doug
Newlin, Guido Schuster, Ikhlaq Sidhu, 3Com; Jiri Matousek, Bay
Networks; Farzi Khazai, Nortel; John Chapman, Bill Guckel, Michael
Ramalho, Cisco; and Chuck Kalmanek, Doug Nortz, John Lawser, James
Cheng, Tung-Hai Hsiao, and Partho Mishra, AT&T.
11. Author's Addresses
Bill Marshall
AT&T
Florham Park, NJ 07932
Email: wtm@research.att.co m
K. K. Ramakrishnan
AT&T
Florham Park, NJ 07932
Email: kkrama@research.att.co m
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DCS Architecture October 1999
Ed Miller
CableLabs
Louisville, CO 80027
Email: E.Miller@Cablelabs.co m
Glenn Russell
CableLabs
Louisville, CO 80027
Email: G.Russell@Cablelabs.com
Burcak Beser
3Com
Rolling Meadows, IL 60008
Email: Burcak_Beser@3com.com
Mike Mannette
3Com
Rolling Meadows, IL 60008
Email: Michael_Mannette@3com.com
Kurt Steinbrenner
3Com
Rolling Meadows, IL 60008
Email: Kurt_Steinbrenner@3com.com
Dave Oran
Cisco
Acton, MA 01720
Email: oran@cisco.com
John Pickens
Com21
San Jose, CA
Email: jpickens@com21.com
Poornima Lalwaney
General Instrument
San Diego, CA 92121
Email: plalwaney@gi.com
Jon Fellows
General Instrument
San Diego, CA 92121
Email: jfellows@gi.com
Doc Evans
Lucent Cable Communications
Westminster, CO 30120
Email: n7dr@lucent.com
Keith Kelly
NetSpeak
Boca Raton, FL 33587
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DCS Architecture October 1999
Email: keith@netspeak.com
Flemming Andreasen
Telcordia
Piscataway, NJ
Email: fandreas@telcordia.com
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DCS Architecture October 1999
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DCS Group Category: Informational - Expiration 4/30/00 19
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