The long-distance Voice over IP (VoIP) network solution is a set of
network design and configuration strategies that provide
trunk-level transport of global switched telephone traffic
distributed over VoIP.
Calls originate in the Public Switched Telephone Network (PSTN),
are routed through interexchange carriers (IXCs), and are handed
off to a wholesale VoIP carrier for transport. To the subscriber,
the long-distaance service seems like any other inexpensive
long-distance service. To the originating
long-distance carrier, the wholesale carrier is only one of a
number of termination options.
The long-distance VoIP network solution offers service providers
the required architecture design, network components, software
features, functional groups, and provisioning methodologies needed
to run a VoIP wholesale service. This solution enables the service
provider to design a wholesale network and sell unbranded voice
services to retailers, such as internet telephony service providers
(ITSPs), application service providers (ASPs), IXCs, and post,
telephone, and telegraph (PTTs) administrations.
This document describes the fundamentals of long-distance VoIP and
provides a nine-step methodology for designing and implementing a
long-distance VoIP network solution. This document is intended to
cover the high-level design of a long-distance VoIP network;
therefore, it does not discuss specific
configuration information.
This document includes the following sections:
• Long-Distance VoIP Network Overview
• Long-Distance VoIP Design Methodology
Long-Distance VoIP Network Overview
The long-distance VoIP network solution includes multiple
components in various combinations from both Cisco and third-party
vendors. Voice points of presence (POPs) that are connected to
other service providers are a central component in the delivery of
wholesale voice services. The types of
interconnections or call topologies service providers support will
determine the specific components and design methods we recommend.
Service providers use the call topologies to build a set of
deployment scenarios that enable wholesale applications. Figure1
shows some of the interconnection
possibilities.
Long-Distance VoIP Network Benefits
The long-distance VoIP network solution provides the following
benefits:
• Voice quality that is indistinguishable from that of the
PSTN
• A cost-effective, reliable VoIP network infrastructure
• Support for least-cost routing and other enhanced call-routing
methods
• Intercarrier call authorization and accounting
(peer-to-peer)
• Support for intercarrier clearing and settlement services
• Support for local, national, and international dial plans
• Connectivity with the PSTN over carrier interfaces
• Connectivity with other VoIP service providers and the VoIP
equipment of other vendors
• A worldwide network of other VoIP service providers interested
in interconnecting
Long-Distance VoIP Design Methodology
To design your own personalized long-distance VoIP solution,
systematically perform the following
Cisco-recommended steps:
Step 1 Identify the services you plan to sell.
Step 2 Identify the type of carriers or providers with which you
plan to interconnect.
Step 3 Determine the interconnection types you plan to use.
Step 4 Determine the call topologies you plan to use.
Step 5 Identify the appropriate deployment scenario.
Step 6 Identify the functional areas you require.
Step 7 Identify the required hardware and software
components.
Step 8 Identify design and scalability issues.
Step 9 Configure and provision components.
Step 1: Identify Services
A key feature of the Cisco long-distance VoIP solution is its
ability to support various mixes of services to suit the needs of a
single service provider or multiple partnering service providers.
Supported services are described in the following sections:
• Minutes Aggregation and Termination (Including ASP
Termination)
• Calling Card Services
• Clearinghouse Services
• Service Options
Minutes Aggregation and Termination (Including ASP
Termination)
The Cisco wholesale voice solution supports the originating carrier
that hands calls over to a VoIP wholesaler at a profit. Termination
settlement rates are generally lower than PSTN termination
rates—the key reason why long-distance carriers will choose a
VoIP carrier for termination.
Furthermore, termination bandwidth is often available over VoIP to
countries where PSTN termination is unavailable because of
congested international gateway facilities or other reasons. The
average call success rate is as good as or better than that
provided by PSTN carriers, and voice quality, including
echo cancellation, is uncompromised.
Key features of this service include the following:
• H.323 VoIP interconnect using standards-based H.323
implementation
• Gatekeeper LRQ forwarding for call routing and accurate call
accounting
• Support for voice, modem, and fax calls
• Support for SS7, T1/E1 E&M, E1 R2, and E1 PRI
interfaces
As part of this service, ASP carrier to carrier termination
services are supported. The ASP originates the call, often over an
Internet-enabled PC-telephony application, or through a PSTN portal
for cellular phone callers. The ASP provides pre-call services,
such as content delivery (prerecorded messages, voice mail, private
number dialing) or supervision-related services, such as “find
me/follow me.” The ASP then hands off any long-distance calls to
a wholesale carrier for termination by the PSTN. This service
requires accurate call accounting.
Calling Card Services
The Cisco wholesale voice solution supports the following calling
card services:
• Prepaid—A wholesale VoIP carrier can host prepaid services
for multiple service providers on its infrastructure. In addition,
most prepaid service providers use VoIP wholesalers to terminate
long-distance calls that are placed by prepaid subscribers. Using
the integrated voice response (IVR) feature in Cisco wholesale VoIP
gateways, and the real-time authorization and call accounting
systems provided by Cisco Ecosystem Partners, you can offer this
service over a VoIP network and lower the cost and deployment time
of calling card services.
• Postpaid—Like the prepaid service, this service can be hosted
by a wholesale VoIP carrier. An example is basic calling that is
accessed by the 800 prefix, a calling card number, or a personal
identification number (PIN). Postpaid service is similar to the
prepaid service, except that with postpaid service the
authorization is not tied to call rating. Consequently, call rating
need not happen in real time, and there may be more partner
billing-system options that perform adequately at scale. After
calls are made, a billing system contracted by the company charges
the carrier.
Clearinghouse Services
When multiple
partners join to provide wholesale voice services, the services
described in the preceding sections may require the assistance of
clearinghouse services for billing and settlement. The Cisco
wholesale voice solution supports call termination agreements
through Open Settlement Protocol (OSP) in Cisco devices.
OSP relies upon Cisco Open Packet Telephony (OPT) framework at the
call control layer. Service providers that use OSP (the only
standard IP interface for VoIP clearinghouse functions), must do
business with only one settlements provider. As a result, there is
no need to negotiate separate agreements with carriers in multiple
countries, meet varied technical requirements for interconnection,
make repeated arrangements for call accounting, or establish
multiple credit accounts. The OSP clearinghouse solution virtually
eliminates the risk of doing business with new service providers
that have a limited credit history—or with carriers in countries
subject to currency fluctuations. In addition, it gives virtually
every VoIP provider the worldwide calling reach it requires.
OSP uses a standard protocol approved by the Internet Protocol
Harmonization over Networks organization of the European
Telecommunications Standards Institute (ETSI TIPHON). By allowing
gateways to transfer accounting and routing information securely,
this protocol provides common
ground between VoIP service providers. Consequently, third-party
clearinghouses with an OSP server can offer call authorization,
call accounting, and settlement—including all the complex rating
and routing tables necessary for efficient and cost-effective
interconnections.
In most cases, a wholesale provider will subcontract with a
clearinghouse partner to provide wholesale voice services with
proper settlement. However, a clearinghouse solutions vendor can
also independently take advantage of the Cisco wholesale voice
solution to achieve market objectives.
Service Options
In addition to the services
previously listed, the other two service options are described in
the following sections:
• Limited Egress Carrier-Sensitive Routing
• Interconnection to Clarent-Based Clearinghouses
Limited Egress Carrier-Sensitive Routing
As an enhancement to simple carrier-interconnection applications,
the Cisco wholesale voice solution makes it possible to route a
call to different destination carriers. You have the same
considerations as with simple carrier-interconnection models, but
with slightly increased call-routing responsibilities.
The directory gatekeeper can make limited egress carrier-sensitive
routing (CSR) decisions by using the sequential LRQ feature, which
is available to the applications using directory gatekeeper
routing.
Generally speaking, this means any TDM partners and directory
gatekeeper peering partners, but also includes any OSP partners in
which an OSP interconnection zone is used, as opposed to a direct
implementation on your gateways.
In this CSR application, the sequential LRQ feature is used to
route a call to different carriers, each of which supports a
different destination. For example, you may provision your
gatekeepers to route certain destination patterns to carrier A
first. If carrier A (an Internet telephony service provider, or
ITSP) is unavailable as a result of a location request reject (LRJ)
or LRQ timeout, you may decide to route the call to carrier B (an
interexchange carrier, or IXC), then to carrier C, and so on.
There are three restrictions to remember with limited egress
CSR:
• Independence of ingress and egress carriers. The egress carrier
is selected independently of the source carrier. The gatekeeper
routes calls on the basis of dialed number identification service
(DNIS). The list of possible egress carriers that you statically
configure are tried in order. Routing decisions are not based on
which carrier sourced the call. For example, the fact that carrier
A sourced the call does not influence the carrier on which the call
will be terminated.
• Independence of destination carriers. Each destination carrier
must be contained in its own zone. For ITSP carriers, this is
fairly simple. Interconnected ITSPs are seen as single remote zones
to which your directory gatekeeper sends LRQ messages. For
interconnected TDM carriers, this implies that the gateways that
can send calls to the carrier are grouped into their own hopoff
zone that is managed by a gatekeeper, and that multiple carriers
are never supported by a single gateway.
• Static versus dynamic routing. Dynamic routing decisions are
not supported; you configure the order of sequential LRQs
statically. Consequently, there is no provision for
percentage-based routing, maximum minute cutoffs, and so on. Egress
carriers are always chosen from a statically configured list of
routes. If the directory gatekeeper determines that an OSP
interconnection zone handles a route, it is possible that the OSP
server returns a terminating gateway on the basis of advanced
routing logic (if so provisioned). For example, the OSP server may
dynamically select a least-cost, terminating carrier on the basis
of time of day or best voice quality.
Interconnection to Clarent-Based
Clearinghouses
You can interconnect with a
Clarent-based service provider (URL www.clarent.com) provided that
the gateways register to a Clarent gatekeeper; however, this would
mean dedicating specific gateways to be part of the Clarent zone.
Back-to-back gateways may be used to provide a “transit” zone
between the Cisco and the Clarent-based network. One of the
back-to-back gateways registers to a Clarent gatekeeper in the
Clarent-based service provider network; the other registers to a
Cisco gatekeeper in your network. This architecture is very similar
to using back-to-back gateways to interconnect OSP partners, except
that here the relationship is H.323 gateway to gatekeeper instead
of OSP.
There are two limitations to using Clarent-based interconnection as
follows:
• IP-to-IP interconnection. The use of back-to-back gateways
enables Clarent-based interconnection partners to exchange traffic
not only with wholesaler TDM-based interconnections, but also with
other IP-based interconnection partners. Those partners may be
either directory gatekeeper- or OSP-based. It may be necessary to
modify the dial plan architecture to support directory
gatekeeper-based IP carrier interconnections.
• Interoperability considerations. Before interconnection is
possible with Clarent-based networks, H.323 interoperability must
be sustained between Cisco gateways and Clarent gatekeepers.
Currently, only voice-bearer interoperability is supported for
G.711, G.723.1, and G.729 codec types. Because of tandem
compression, back-to-back gateways impair voice quality.
Step 2: Identify Carriers or Providers
As a wholesale voice service provider, you need to interconnect
with other service providers (ITSPs and ASPs) and carriers (IXCs
and PSTNs) in order to offer the services you selected in Step 1.
This interconnection method is referred to as a call topology.
Because each call topology is specific to the
carrier or service provider with which you plan to connect, you
need to first identify the targeted carriers and service
providers.
