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Table Of Contents
Cisco Signaling Controller Product Information
Signaling Controller Management
Cisco Signaling Link Terminal Management
PSTN Voice and Fax Traffic Offload
Packet Telephony and Data Wholesale Services
Introduction
This chapter introduces the Cisco SS7 Interconnect for Voice Gateways Solution and includes the following sections:
• Overview
• Cisco Signaling Controller Product Information
• Benefits
• Features
Overview
When you make calls using your telephone or PC with modem connection, you generally use the Public Switched Telephone Network (PSTN). You hear progress tones such as a dial tone or a busy tone that tell you your call is being processed. These are alerting tones that are produced by the PSTN. Alerting tones are used to communicate from the PSTN to users, and telecommunications computers, or "switches," communicate with each other through standards-based signaling. Signaling is the backbone for interconnection between carrier, cellular, and wireless networks. Signals are the medium that set up and tear down your calls.
A major breakthrough in signaling networks was to separate the signaling path from voice and data conversations. This is called "common channel interoffice signaling (CCS)." CCS, also known as out-of-band signaling, is a data network that overlays a carrier's switching network. Using CCS increases network intelligence, efficiency, automation, and functionality. CCS has evolved into a standard called Signaling System 7 (SS7), a protocol that lowered costs and increased network reliability even further. With SS7, all carriers are able to interoperate as a consistent and seamless network. Services such as global billing, wireless roaming, and 800 number calling rely on the SS7 protocol to exchange messages reliably.
The Cisco SS7 Interconnect for Voice Gateways Solution is a distributed system that provides SS7 connectivity for Voice-over-IP (VoIP) access gateways by using the Cisco Signaling Controller (also referred to as the Cisco SC2200 product) and the access gateways as a bridge from the H.323 IP network to the PSTN network. This solution interacts over the IP network with other Cisco H.323 VoIP access gateways. In addition, the Cisco SS7 Interconnect for Voice Gateways Solution can interoperate with H.323 endpoints, using non-SS7 signaling such as ISDN PRI and channelized T1.
Figure 1-1 illustrates where the Cisco SS7 Interconnect for Voice Gateways Solution is located when it is dropped into a PSTN to offload calls. By placing the solution as close to the ingress switch as possible, voice and data traffic ties up fewer PSTN resources. The direct connection of the Cisco SS7 Interconnect for Voice Gateways Solution to the SS7 network provides advantages such as faster call setups and teardowns, and SS7's look-ahead capabilities for rerouting to avoid downed network nodes and links.
Figure 1-1 With a Cisco SS7 Interconnect for Voice Gateways Solution
Cisco Signaling Controller Product Information
The Cisco signaling controller (Cisco SC2200) product is an application supported by the Cisco media gateway controller (Cisco MGC) product line that addresses the scalability of ISP dial modem traffic. The Cisco MGC is a powerful call control application that can be used to facilitate a wide variety of telecommunication services using data infrastructures. This new world architecture allows end users to realize the benefits of deploying a single network to address both data and voice applications.
Note Your Cisco SS7 Interconnect for Voice Gateways Solution documentation suite includes the Cisco MGC reference books.
Note Some product labels and packaging might use the term Cisco telephony controller. Any references to the Cisco telephony controller apply to the Cisco media gateway controller.
Understanding Terminology
The key terms used to describe the Cisco SS7 Interconnect for Voice Gateways Solution architecture are:
•Cisco SC2200—A hardware and software package that provides the signaling controller function. Typically this includes two SC hosts configured in a redundant manner for increased availability.
•SC host—A Sun host running signaling controller software.
•SC node—The combination of hardware (Sun servers and Cisco SLTs) and software that provides the signaling controller function and transports the signaling traffic between the SC hosts and the SS7 signaling network.
•SC zone—The combination of an SC node and the Cisco access gateways that are provided with signaling services.
Architecture
The Cisco SS7 Interconnect for Voice Gateways Solution architecture provides SS7 connectivity for voice gateways, by way of the SC host as a protocol translator, using ISDN Q.931 and SS7 as control protocols.
The solution consists of the following required components that are described in more detail in the "Cisco SS7 Interconnect for Voice Gateways Solution Components" section of this document.
Figure 1-2 Cisco SS7 Interconnect for Voice Gateways Solution Architecture
Benefits
Using the Cisco SS7 Interconnect for Voice Gateways Solution provides the following benefits:
•Provides wholesale dial services, dial-up Virtual Private Networks (VPNs), Internet access, and voice services, while interconnecting as a carrier.
•Provides network interface between the PSTN and the H.323 network.
•Addresses voice network congestion by using the SS7 interfaces to make features such as rerouting on overflow conditions and the use of IN functions possible, which further drives down operating costs.
•Replaces ISDN PRIs with bearer trunks and increase available bandwidth.
•Installs an SS7 POP in a new location without the added expense of a switch.
•Integrates the access gateways directly into the SS7 network, using the Cisco SS7 Interconnect for Voice Gateways Solution, and thus remove the need for two switch ports on the PSTN circuit switch for each NAS port installed.
•Increases the signaling channel to bearer channel ratios, thus decreasing the number of signaling channels needed and the overall complexity of the system or network.
•Provides economical reliability of SS7 link termination by using channelized T1/E1/T3 software on the Cisco SLT.
•Transparently passes individual and uncompressed T1/E1/T3 channels between T1/E1/T3 ports using Drop and Insert interfaces.
•Provides enhanced scalability of PSTN and packet network interconnectivity.
•Provides enhanced usability of the Cisco SC2200 system utilities.
Features
Table 1-2 briefly lists features that are provided with your Cisco SS7 Interconnect for Voice Gateways Solution. For an overview of scalability and performance, system redundancy, management, and software requirements, see subsequent sections of this document.
Scalability and Performance
The Cisco SS7 Interconnect for Voice Gateways Solution includes the following scalability and performance features:
•Support for up to 100,000 DS-0 ports
•Ability to process 80 calls per second with 100,000 simultaneous calls of 20 minute call hold time
•Support for up to 180,000 busy hour call attempts
•Support for 250+ destination point codes (DPCs)
•Support for 6 originating point codes (OPCs)
•Support for quasi-associated or fully associated signaling
•Complete continuity check (two-wire and four-wire)
•Compliant with NEBS Level 3 standards
System Redundancy
For maximum reliability and resilience, Cisco recommends the following options:
•Deploying the Cisco SS7 Interconnect for Voice Gateways Solution continuous service configuration at your site. The continuous service configuration consists of an active server and a standby server, linked by a heartbeat function. All configuration changes made to the active server are replicated on the standby server.
•Using a minimum of two LAN switches from the Cisco Catalyst switch family that support:
–Inter-Switch Link (ISL) trunking protocol configured between the two switches.
–One route-switch module (RSM), which routes traffic between the VLANs when necessary.
•Using a minimum of two links per linkset if signaling links are connected to the Cisco signaling controllers. The links should be split across separate T1/E1 interface cards on the LAN switches, SC hosts, and Cisco SLTs.
Signaling Controller Management
Table 1-3 provides an overview of the management components of the signaling controller.
Access Gateway Management
The Cisco IOS software installed on the access gateways provides an array of network management components (described in Table 1-4) designed to meet the needs of today's large, complex networks.
These management features do the following:
•Reduce network bandwidth and processing overhead
•Offload management servers
•Conserve resources
•Ease system configuration tasks
Cisco's integrated management simplifies administrative procedures and shortens the time required to diagnose and fix geographically dispersed networks with a small, centrally located staff of experts. Configuration services reduce the cost of installing, upgrading, and reconfiguring network equipment.
Cisco Signaling Link Terminal Management
The Session Manager software, running on the Cisco SLT, manages the communication sessions between two SC hosts.
The session manager:
•Maintains separate communication sessions with each signaling controller in the pair
•Uses RUDP to communicate between the Cisco SLT and the signaling controller
•In a continuous service configuration, handles additional traffic in the event of a single Cisco SLT failure, with no impact to call processing
Applications
The Cisco SS7 Interconnect for Voice Gateways Solution preserves full capabilities on the IP side, including existing H.323 access gateway functionality, interoperability with H.323 gatekeepers, two stage dialing, mixed gateway environment, and PSTN voice and fax traffic offload. This solution targets the following key applications that support phone to phone, PC to phone, and fax relay packet telephony services:
• PSTN Voice and Fax Traffic Offload
• Packet Telephony and Data Wholesale Services
Settlement providers require services such as two-stage dialing, international toll bypass, and open settlements, as illustrated in Figure 1-3.
Figure 1-3 A Typical Settlement Provider Application
Two-Stage-Dial Toll Bypass
This application enables the service provider to leverage its WAN infrastructure and offer long-distance toll bypass services. Each customer is assigned an account number and a personal identification number (PIN). The user dials a local or a 1-800 Internet Telephony Service Provider (ITSP) number and is connected to the local VoIP point of presence.
In Figure 1-3, the first leg of the call is terminated by the access gateway through the ISUP to Q.931+/IP translation. The user is prompted by the interactive voice response (IVR) to input account and PIN numbers, and (following an authentication by the RADIUS server) then a secondary dial tone allows the desired phone number to be entered. Also, the RADIUS authentication can be accomplished based on the combination of calling number ID (ANI) and called number (DNIS).
For fax services, the PIN number is automatically dialed by the redialer that is attached to the fax machine. An E.164 destination phone number, dialed by the user, is mapped by the local-zone gatekeeper to an IP address of a remote-zone gatekeeper. The remote-zone gatekeeper selects an access gateway to terminate the call. During the call setup, end-to-end ISUP signaling transparency is supported. After the call setup is complete, the voice is encoded by the access gateway using standard algorithms (including G.711, G.729, G.729a, and G.723.1), encapsulated in Real-Time Transport Protocol (RTP) packets, and then routed over the WAN to the remote access gateway that decodes the voice and delivers it to the receiver.
PSTN Voice and Fax Traffic Offload
This application offloads PSTN traffic from congested PSTN networks. The offloaded traffic is then forwarded to a tandem switch connected in a peer-to-peer configuration to the Cisco SS7 Interconnect for Voice Gateways Solution through access gateways. The configuration uses Direct Inward Dialing (DID) functionality, and no ANI or CDR billing is required.
Packet Telephony and Data Wholesale Services
This application enables the wholesale service provider to extend services to a specific geographical area. Again, voice and fax traffic is offloaded from PSTN onto data networks, as well as support scalable routing and Quality of Service (QoS), across data networks, to effectively route the calls to their final destination. Routing and authorization between multiple service carriers, and real-time selection of a cost-effective service provider and reciprocal billing arrangements are required. Resource pooling is required for data, but not for voice.
The Open Settlement Protocol (OSP) is a communications protocol between H.323 equipment and back-end services, such as authorization, route determination, and call detail record (CDR) collection and settlement. With OSP, interdomain CDR records are forwarded to the settlement system, processed, and then transmitted as settled records to billing system platforms. The access gateways in your Cisco SS7 Interconnect for Voice Gateways Solution generate intradomain records and are accessed by any accounting and billing system based on RADIUS.
Posted: Thu Oct 14 10:11:24 PDT 2004
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