A critical part of the core infrastructure for packet voice networks is its ability to interconnect with the existing time-division multiplexing (TDM) networks. Call setup, progress, and teardown are all accomplished via complex network signaling commands that traverse the network end to end.
While several signaling options exist, most new telephony services depend on the power and reliability of Common Channel Signaling System 7 (SS7/C7), more commonly abbreviated and known as SS7. Differing from in-band signaling methods such as R2 or Channel Associated Signaling (CAS), an SS7 architecture dictates that call control information traverse a separate network dedicated solely to the business of transmitting the required information to successfully connect telephone calls.
Cisco SS7 Interconnect for Voice Gateways is a network option under the Cisco Voice Infrastructure and Applications (VIA) solution. The Cisco VIA solution provides an architecture that is well suited for future voice-over-IP (VoIP) network expansion, and the Cisco SS7 Interconnect option allows customers to exploit the many advantages of introducing SS7 signaling into their VoIP networks. Using Cisco SS7 Interconnect, new and existing Cisco VIA customers will immediately realize a reduction in one-time and recurring facilities costs.
The Cisco VIA solution is a proven infrastructure; it uses standards-based protocols to enable a broad portfolio of packet-based voice services. It is designed for service providers and international carriers who seek to increase revenue, build customer loyalty, and boost profits by adding national and international call transport and a variety of other services such as prepaid and postpaid calling card services, voice mail and unified communications, application service provider (ASP) termination services, and dial access to their service portfolios. Support for industry-standard protocols such as H.323 and Session Initiation Protocol (SIP) make it easy to link a service provider's network through a broad number of interconnect options.
The Cisco VIA solution is a mature, time-proven solution, first deployed in 1998 and now installed in more than 100 customer networks in more than 80 countries. Cisco provides 24-hour online and phone support for Cisco VIA customers through the Cisco Technical Assistance Center (TAC). In addition, all components in the Cisco VIA solution, including the Cisco SS7 Interconnect option, are extensively tested in a solution environment, which ensures reliable operation and seamless interoperability with other networks and network components.
One of the many advantages of the Cisco VIA solution is its modular approach to providing services. Service providers can size their network and implement network applications that enable services based on their current business model requirements, and then add infrastructure and applications as future requirements dictate.
The Cisco VIA solution begins with a powerful core infrastructure layer that provides transport, routing, and signaling across enterprise and service provider networks. The flexibility of the core infrastructure permits service providers to choose among signaling and routing options for the services they wish to offer today. They can also add to their core infrastructure as their networks grow—without a complete hardware upgrade—and add to their portfolio of voice and data services.
Building upon the core infrastructure layer, the Cisco VIA solution services layer enables service applications that help increase revenue. The services layer also enables service providers to build access networks that extend their core infrastructure. Figure 1 illustrates the relation between the services layer and the core infrastructure layer, showing the revenue-generating services that can be offered over a Cisco VIA network.
Cisco VIA Solution Modules
As an infrastructure option of the Cisco VIA solution, Cisco SS7 Interconnect enables SS7 as a signaling method in the core infrastructure of the VoIP network and underpins baseline service applications currently offered as part of the Cisco VIA solution.
The Cisco VIA solution enables a host of services such as national and international transport, prepaid and postpaid calling card services, ASP termination services, voice mail and unified communication services, and dial access. All of these service applications are fully compatible with the Cisco SS7 Interconnect option.
The Cisco VIA solution allows service providers to take advantage of packet network investments. With the Cisco VIA solution, service providers can quickly build multiservice networks and deploy revenue-generating, converged voice-and-data services. The Cisco SS7 Interconnect option builds upon this opportunity by expanding available interconnect options and lowering operating costs. Key benefits to customers include:
- Reduced recurring charges—In general, tariffs favor interconnection using SS7 signaling. Intermachine trunks (IMTs) are less expensive than ISDN-based facilities, both on a one-time (installation and provisioning) and recurring (monthly charge) basis. This equates to lower monthly expenses, reduced cost of goods sold and higher margins for service providers.
- Opportunity to enter new markets—Many countries will not permit alternate carriers such as competitive local exchange carriers and Internet telephony service providers (ITSPs) to interconnect to the public switched telephone network (PSTN) with signaling methods other than SS7. The Cisco SS7 Interconnect option allows Cisco VIA customers to enter those markets that may have been previously closed.
- Network scalability—SS7 signaling offers superior scaling over other signaling methods such as R2, Primary Rate Interface (PRI), and CAS.
- Backwards Compatibility—Cisco SS7 Interconnect is backwards compatible with all previous Cisco VIA signaling methods. Customers can rest assured that revenue-generating services supported by their VoIP network will continue to operate and be fully supported with the addition of the SS7 signaling option. Also, upgrading to the Cisco SS7 Interconnect option is easy and straightforward. With the addition of the Cisco PGW 2200 PSTN Gateway and the Cisco Signaling Link Terminal (SLT), SS7 appears as simply another signaling method to the VoIP network.
- Faster customer response time—Telecommunications operations tend to be optimized around ISDN User Part (ISUP) trunk IMT provisioning. Therefore, IMTs can generally be provisioned more quickly than PRI, CAS, or R2 trunks. SS7 enables service providers to contract for IMTs as opposed to PRIs, CAS, or R2 trunks.
- Widespread network ubiquity—SS7-signaled trunks are usually supported at all TDM switch and point-of-presence (POP) sites. Other trunks, especially PRI-signaled trunks, may only be supported on a subset of PSTN switches.
The Cisco VIA solution architecture has four fundamental hardware elements: a gateway, a gatekeeper (for H.323 networks), a SIP proxy server (for SIP networks), and a billing mediation server. Collectively, these products provide PSTN-to-VoIP network access, security, call detail records (CDRs), resource management, and call routing.
The Cisco SS7 Interconnect option adds the Cisco PGW 2200 PSTN Gateway to the voice network. There are two additional hardware elements: the Cisco Signaling Link Terminal (SLT) and the Cisco PGW 2200 hosts. The Cisco SLT platform terminates the physical links between the SS7 network and the Cisco PGW 2200. In signaling mode, the Cisco PGW 2200 provides SS7 signaling for H.323 and SIP-based VIA networks, interpreting call signaling information that is used to establish and tear down calls. When configured in signaling mode, the Cisco PGW 2200 interfaces with the gateways using the Cisco extended Q.931 protocol.
In call control mode, the Cisco PGW 2200 performs call control tasks such as digit analysis, circuit selection, complex and conditional routing, and Intelligent Network triggers. When provisioned in call control mode, the Cisco PGW 2200 interfaces with the gateways using Media Gateway Control Protocol (MGCP).
Table 1 Cisco VIA Solution and Cisco SS7 Interconnect Components
Figure 2 depicts the Cisco VIA solution, including the Cisco SS7 Interconnect option, in an architecture diagram. As an integral part of a Cisco VoIP infrastructure, Cisco SS7 Interconnect can be used to meet a customer's SS7 signaling requirements whether that customer seeks to offer transport services, ASP termination services, or business access. It supports the most commonly used standards-based packet telephony signaling methods such as H.323 and SIP protocols.
Cisco VIA and Cisco SS7 Interconnect Architecture
The Cisco SS7 Interconnect option uses a unique, distributed architecture to terminate SS7 signaling in the Cisco VIA network. The Cisco PGW 2200 consists of two components, the Cisco Signaling Link Terminal (SLT) and the Cisco PGW 2200 hosts, which share the responsibility of processing first lower- then upper-layer SS7 messaging.
The Cisco PGW 2200 is a versatile call control platform providing SS7 interconnection for H.323 and SIP networks. The Cisco PGW 2200 supports both call agent and Cisco IOS based call models for both voice and dial applications.
The Cisco PGW 2200 analyzes signaling received from connecting entities over the IP signaling backhaul network, finds needed resources to process calls, analyzes service requests, and sends execution commands to media gateways that provide ingress and egress to the VoIP packet core. It also interfaces with the PSTN via standard Transaction Capabilities Application Part (TCAP) and Intelligent Network Application Protocol (INAP) interfaces to provide access to Intelligent Network and Advanced Intelligent Network (AIN) platforms and services.
A core component in the Cisco SS7 Interconnect option, the Cisco PGW 2200 provides SS7 signaling support for H.323- and SIP-based networks in a call model based on Cisco IOS Software. It communicates with the media gateways via the Cisco extended Q.931 call control protocol. Call routing decisions are determined on a gateway-by-gateway basis, under the control of gateway-based dial peers, H.323 gatekeepers, or SIP proxy servers. Policy-based routing is available with the Cisco Carrier Sensitive Route (CSR) Server.
- Continuity testing—Both loopback and transponder continuity testing options are supported, adjustable on a percentage-of-call basis.
- Calling and called number analysis—Both "A" number and "B" number analysis are supported, including many individual capabilities for each.
- Blacklist and whitelist screening—Calls can be either accepted (whitelist) or denied (blacklist) based on number analysis and list comparison.
- Continuous operation—Offers stable call preservation during catastrophic hardware or software failures.
- Overload control—Maintains machine congestion levels, an internal measurement of its own current congestion level. There are four user-configurable congestion levels that alter the behavior of the Cisco PGW 2200 in order to process the maximum level of calls while in overload conditions.
- Local number portability—Support for this feature allows end users to change service providers without changing their phone number.
- Open software—All call control on the Cisco PGW 2200 is enabled using the Cisco Media Gateway Controller (MGC) software running on UNIX open computing platforms. The software is designed to deliver 99.999 percent, or "five nines" availability.
One of the strengths of the Cisco PGW 2200 is its support of a large variety of SS7 ISDN System User Part (ISUP), Telephone User Part (TUP), and National User Part (NUP) variants used globally by carriers and telephony switches. The Cisco PGW 2200 is certified in more than 50 countries, and currently has a portfolio of more than 90 variants.
For additional information on the Cisco PGW 2200, please refer to the Cisco.com product Web site at:
"A" or "F" links from the SS7 network are physically connected to the Cisco Signaling Link Terminal (SLT) via one of several supported interface cards. The Cisco SLT terminates Message Transfer Part (MTP) Layers 1 and 2 of the SS7 protocol stack. As MTP 2 is a message and processor-intensive layer of SS7 signaling, terminating it on the Cisco SLT frees the Cisco PGW 2200 from wasting cycles on lower-layer functions.
- Reduced equipment costs—The Cisco SLT uses generally available Cisco platforms. There is no custom-built hardware required, so costly application-specific equipment that tends to drive up costs can be avoided.
- Wide variety of facility interface cards—The Cisco SLT offers a number of different interface cards for SS7 circuit termination. The Cisco SLT can terminate at 56 kbps, T1, and E1 SS7 facilities.
- Colocated or remote deployment options—Cisco SLTs can be deployed in either colocated or remote configurations. In a colocated configuration, the Cisco SLT is physically located with the Cisco PGW 2200 host controller and is connected to the call control platform via a LAN. In remote deployments, the Cisco SLT is physically located somewhere other than with the Cisco PGW 2200 (generally with the signal transfer point [STP] or service switching point [SSP]), with SS7 signaling being backhauled to the call control platform via an IP-based WAN.
The Cisco SLT is available in both standalone and integrated platforms. The standalone version is supported on Cisco 2611 and 2651 multiservice platforms. The integrated SLT is available as an option on Cisco AS5350 and AS5400 universal gateways.
When the Cisco SS7 Interconnect option is used in Cisco VIA networks, the solution will support common signaling interconnect methods as well as features that exploit the power and savings of IP-based networks.
Access or "A" links are used between the SSP (served by the Cisco PGW 2200) and the STP to connect the Cisco VIA network to the PSTN. These links are dedicated to signaling, meaning that no bearer traffic is provisioned on the facility, even if there are available time slots not being used by signaling. For reliability, there are generally, at a minimum, two "A" links provisioned between the solution and the home STP pairs. While most commonly seen in North America, "A" links are used in other geographic areas such as Asia and South America, depending on availability of facilities.
Fully associated or "F" links are generally deployed in the Cisco VIA solution when a large volume of traffic exists between two SSPs (between the Cisco PGW 2200 and a PSTN switch, that is), or when it is not feasible to connect directly to an STP. "F" links are not commonly used in North America, but they are widely deployed in Europe, Asia, and South America. While "F" links can be provisioned with only signaling on the facility, the most common configuration includes the provisioning of bearer channels on time slots not used by SS7 signaling.
The "drop and insert" application feature (part of the Cisco SS7 Interconnect option), also known as time-division multiplexing (TDM) cross-connect, allows Cisco VIA customers to deploy "F" links with both signaling and bearer traffic on the same facility. The "drop and insert" feature grooms the signaling channels from the facility and backhauls the signaling to a Cisco PGW 2200 for processing. Bearer channels are "hairpinned" on the Cisco SLT interface card and sent to a voice gateway by connecting a cable between the egress port on the Cisco SLT voice WAN interface card (VWIC) and an available T1 or E1 port on a gateway.
The integrated SLT offers internal drop and insert, which requires no external cabling. All bearer channels are internally hairpinned on the gateway and sent to the VoIP network. Figure 3 depicts both the Cisco SLT and integrated SLT "drop and insert" architectures.
"Drop and Insert" Architectures: Cisco SLT and Integrated SLT
The Cisco VIA solution supports both colocated and remote termination of SS7 signaling links. With colocated configurations, the Cisco SLTs are physically located with the Cisco PGW 2200, and are connected to the host via a LAN. Remote configurations are defined as deployments where the SS7 signaling is backhauled over an IP-based WAN from an SSP or STP to the Cisco PGW 2200. Figure 4 illustrates this concept.
Remote SS7 Signaling Backhaul
In many countries, the cost of dedicated signaling links can be prohibitive. Dedicated signaling links may also be unavailable due to facilities shortages. Customers can now take advantage of IP-based WANs to backhaul SS7 signaling from STPs and SSPs to the Cisco PGW 2200.
Part of the Cisco distributed SS7 architecture involves the transport of SS7 signaling information over IP networks. The Cisco SLT, using the Cisco Reliable User Datagram Protocol (RUDP), backhauls the upper-layer SS7 protocols across an IP-based signaling control network to the Cisco PGW 2200. Cisco RUDP is a simple, connection-oriented, packet-based transport protocol based on RFC 908 (Reliable Data Protocol) and RFC 1151 (Version 2 of the Reliable Data Protocol).
Cisco Session Manager software manages the communication sessions with the Cisco PGW 2200. When the Cisco SLT is used with the redundant pair of Cisco PGW 2200 host controllers, Cisco Session Manager software maintains separate communication sessions with each controller in the pair. The session between the Cisco SLT and the active controller transports the SS7 traffic, while the session between the Cisco SLT and the standby controller provides backup. Cisco Session Manager software uses Cisco RUDP to communicate between the Cisco SLT and the Cisco PGW 2200.
SS7 comprises a signaling message set that is rich in information. SS7 signaling information is used not only for call establishment and teardown, but also for billing and call routing functions. However, integrating support for SS7 signaling in distributed packet telephony networks poses unique challenges.
The two primary VoIP signaling protocols used today, H.323 and SIP, do not support all the call control elements found in SS7 signaling. While transport of ISUP parameters in packet networks is covered by standards such as H.323 Annex M and SIP-T, these standards call for simple encapsulation and blind transport ISUP call information, and ignore the importance of ISUP parameters with respect to call routing and billing. Simply encapsulating and blindly transporting ISUP call information from one end of the network to the other gives no consideration to nodes within the packet network that may need this information to effectively route calls.
Generic Transparency Descriptor (GTD) mitigates this problem by normalizing ISUP messages and parameters in a generic fashion and using the underlying VoIP call signaling messages to transport the information to nodes within the network. This allows distributed call control to use the information as needed to perform call routing. This approach also allows nodes within the network to modify ISUP call parameters, making them available for advanced call services.
- ISUP transparency and interworking—GTD enables the transport of ISUP messages and parameters in a generic format between VoIP ingress and egress points. The Cisco PGW 2200 interworks GTD and the ISUP variant being used.
- ISUP to ISUP hairpinning—Using GTD, call hairpinning, also known as "gateway TDM switching," permits intragateway call transport back out to the PSTN without losing any critical ISUP messages or parameters.
- R2 to ISUP interworking—GTD enables the transport of generic signaling information between a Cisco PGW 2200 and gateways with R2 signaling interfaces. The gateway provides the interworking function between R2 signals and GTD.
- Transport ISUP parameters to gateway—ISUP parameters can be used by the gateway for call routing, IVR, and certain supplementary services. Parameters of interest include redirecting number, redirecting reason, hop count, and original number called.
- Delivery of ISUP parameters to the RADIUS server for billing—GTD enables the transport of ISUP messages and parameters to the gateway, and subsequent delivery to the RADIUS server. Parameters that can be used by the RADIUS server for billing include calling party category and originating line indicator.
- Advanced call routing—Using GTD, ISUP call messages and parameters can be delivered to a gatekeeper or route server for enhanced call routing. Calling party category, called number with nature of address, forward call indicators (end-to-end method indicators), interworking indicators, and ISUP part indicators are some of the parameters that can be used for advanced call routing in VoIP solutions.
Figure 5 shows the interfaces that transport GTD within an H.323-based Cisco VIA architecture with the Cisco SS7 Interconnect option. All major components (or nodes) within the network are enabled with the GTD builder-parser, and can take advantage of the information transported by GTD.
GTD Interfaces in an H.323-Based Cisco VIA Network
Each of the nodes in Figure 5 uses a distinct signaling protocol in which GTD messages can be encapsulated. Table 2 correlates each GTD-enabled node in an H.323-based Cisco VIA network with its associated signaling protocol. Gateways that support GTD include the Cisco AS5000 family of universal gateways. Gatekeepers include Cisco 3660 and Cisco 7200 series routers.
Table 2 GTD-Enabled Signaling Protocols and Interfaces
Generically, GTD defines three distinct "node types" within a VoIP network. Depending on its location and defined function within the network, each node type displays a particular behavior with respect to its use of GTD. Table 3 describes node types and their functions. See above for examples of Cisco VIA components that may serve as GTD nodes.
Table 3 GTD Node Types and Their Functions
Building upon the information in Tables 2 and 3 above, the function of each GTD-enabled node in a Cisco VIA network is defined in Table 4 with respect to how each node uses the builder-parser to impart signaling information into the network. Table 4 shows Cisco VIA components, their GTD node type, and their role in building and parsing GTD messages.
Table 4 GTD Functions for ISUP Signaling Transparency
GTD technology is currently supported in SS7-enabled Cisco VIA networks using the H.323 signaling protocol. SIP signaling will use GTD technology to support ISUP signaling transparency in subsequent releases of the Cisco VIA solution.
The Cisco VIA SS7 Interconnect option supports advanced call features needed to process and route calls through the VoIP network at a lower overall cost to service providers. As more and more minutes traverse packet networks, Cisco SS7 Interconnect capabilities can be added as networks grow and call processing and routing complexity increases.
While basic call routing can be achieved using gateway-based dial peers and H.323 gatekeepers, many service providers need an additional level of call routing control. The Cisco Carrier Sensitive Route (CSR) Server introduces call routing algorithms to distributed call control networks via a high-performance, low-cost route server. The Cisco CSR Server supports the most common call routing algorithms, including:
- Least cost—Calls are routed through the VoIP network and delivered to the egress point based on the least cost call path.
- Time of day—The call is routed to the egress point based on the time of day.
- Percentage-based—Calls entering the network are routed to trunk groups or carriers based on a percentage allocation.
- QoS—Call paths can be determined based on a predefined QoS through the VoIP network.
The Cisco CSR Server also features an open database interface for ease of integration with a service provider's back-office systems. Well-defined and normalized datasets support direct import via Structured Query Language (SQL) into a relational database.
A trunk group represents the logical grouping of physical interfaces containing similar characteristics on a gateway. More than one interface on a given gateway can belong to the same trunk group if it shares similar characteristics with other interfaces belonging to the same group. Trunk groups can be used to identify the source or destination of calls using PSTN-facing ports. It is also possible for more than one trunk group to be present in an individual gateway.
- Assign trunk or carrier IDs to groups of circuits on gateways
- Group interfaces on a given gateway to form logical trunk groups
- Route calls based on trunk group labels
- Apply call translation rules to trunk groups based on trunk group IDs
- Apply circuit selection algorithms to trunk groups based on trunk group IDs
- Report the trunk group label in the CDR, permitting it to be pushed to the RADIUS server for use in billing
In the PSTN, the operation of trunk groups and the circuits contained within those trunk groups can be classified according to two different ways of controlling seizures for individual calls. Two-way trunks can be seized at either end of the network, while one-way trunks can be seized at only one end of the network.
When one-way trunks are provisioned, the trunk group is usually partitioned into one group that can be seized at one end of the network, and another group that can be seized on the opposite end of the network. Two-way trunk groups are more flexible in servicing fluctuating traffic, but are more difficult to control because the possibility of simultaneous seizures, or "glare," can occur at either end of the circuit. Glare conditions must be resolved in order for the call to proceed.
In this release of the Cisco SS7 Interconnect option for the Cisco VIA solution, support for the most common PSTN circuit selection methods and trunk group algorithms is available. This feature of the Cisco SS7 Interconnect substantially reduces the chance for glare conditions, and dramatically improves the chances of successful call completion. This is especially important as call volume (minutes) increases on the VoIP network.
- Even up—Select an idle circuit with an even timeslot ID from the trunk group in ascending order
- Even down—Select an idle circuit with an even timeslot ID from the trunk group in descending order
- Odd up—Select an idle circuit with an odd timeslot ID from the trunk group in ascending order
- Odd down—Select an idle circuit with an odd timeslot ID from the trunk group in descending order
- Both up—Select an idle circuit from the trunk group in ascending order
- Both down—Select an idle circuit from the trunk group in descending order
- Least idle—Select an idle circuit from the trunk group with the least idle time
- Longest idle—Select an idle circuit from the trunk group with the longest idle time
- Random—Select an idle channel at random within the trunk group
- Sequential—Select a trunk group circuit in sequential order when a free channel is available
Circuit selection algorithms and selected methods are applied on a per-gateway basis and do not span across multiple gateways. This feature works with SS7 and other gateway-supported signaling protocols, including CAS and R2.
- Worldwide support—With both online and phone support, the Cisco Technical Assistance Center (TAC) is available to assist customers with technical issues on a 24-hour basis. The Cisco TAC offers a wide range of service agreements that are tailored to meet the level of support required by both service provider and enterprise customers.
- Long-term commitment to IP telephony—The market leader in VoIP, Cisco has the financial strength, depth of technology, and market presence to endure market cycles and support service provider and enterprise customers for the long term.
- Comprehensive product portfolio—Customers can obtain all the necessary components for the solution—from the central office to the customer premises—from a single source.
To learn more about the Cisco VIA solution and the Cisco SS7 Interconnect for Voice Gateway option, please visit www.cisco.com/ go/telephony.