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March 2007

Features

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Spotlight Events

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For additional support tips, visit the Technical Notes section of the NMS web site.

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For detailed information on NMS’s participation in industry events, conferences, and seminars, visit the Events Calendar.

Features

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Customer News

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Spotlight Events

 

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Telephony Web Services


 By Brough Turner, SVP & CTO, NMS Communications

The following article was featured in the March 2007 issue of Internet Telephony Magazine (www.tmcnet.com).

Today, most major Internet applications provide public programming interfaces. Mapquest introduced an API in 2004. Then when Google introduced their Maps API in June 2005, there was an explosion in applications called Mashups. Mashups integrate data from other sources into Google maps, or integrate Google maps into other web services. Today it’s hard to surf the web without running into 3rd party applications that leverage map services. But maps are just one highly visible example of the burgeoning field of mashups based on web services APIs. Amazon, Flickr, eBay, YouTube, Facebook all offer APIs. These APIs benefit consumers by allowing entirely new services to be easily concocted while helping service providers grow their communities of users (and their ad revenue!).

But parallels in the telephony world are extremely limited. Indeed, using Google to search for the phrase “telephony web services” finds only one of the companies discussed later in this article. This is a nascent field.

Traditional fixed and mobile operators still run highly constrained voice networks. Yes, you can have your computer place and receive calls, but if you want to overlay a private numbering scheme for use between phones in your far-flung offices (a voice VPN) — such services are only available from your operator. Likewise, if you want to integrate your customer care software with call center functionality, you start with PBX technology, as public networks are not open. Even access to your own call history requires scraping data from itemized bills.

Does the new world of VoIP do any better? Not much. Major VoIP operators, like Vonage, use VoIP to reproduce traditional fixed-line services — basically “digital POTS” — no open APIs there. Skype offers PC-based APIs for the Skype client and you could argue that’s all that makes sense for a peer-to-peer service. There are no central services to expose through web service APIs. But the test is what can you do with their APIs and the answer is client extensions. Since Skype clients run as a single instance per PC, you can’t easily implement a multi-channel back-to-back user agent (B2BUA) and a VPN.

Telephony Web Services Technology

The good news is a good number of companies are developing telephony web services technology and a few offer hosted services directly or through partners. None of this is free and no service is widespread, but it is a sign of things to come.

LignUp Corporation, Abbeynet S.p.A., Ubiquity Software and Angel.com offer SIP-based VoIP service platforms that can be controlled via web services APIs and they each allow access at several levels including some fairly high-level functionality. LignUp offers basic media server and call control functionality but also complete PBX and messaging systems that are configured and controllable via web APIs. Angel.com is more focused on call center applications while Ubiquity offers conferencing and a complete residential VoIP service package. Abbeynet also focuses on VoIP services, with an optional pre-paid billing platform and a presence server.

As an example of what is possible, LignUp did a mashup using their telephony services API and the popular hosted customer relationship management system, Salesforce.com. A significant problem with any sales force automation system is getting field sales to actually input data. The Salesforce.com telephony mashup took two days to implement and the result is slick, handling everything you’d expect, including calling customers, capturing telephony information and filling in Salesforce.com forms for both incoming and outgoing calls.

Each of the platform vendors has companies providing hosted services on their platform. For example, Streamdoor Ltd. provides hosted contact center services on a LignUp platform. It’s configured and controlled via web service APIs, but the work is principally by Streamdoor on a customer-specific basis.

The closest I’ve found to customers doing their own mashups is with hosted service provider M1 Global. They focus on business process automation for enterprise customer care, i.e. the intersection of business policies and procedures with call centers and front line employees. They provide not only web service APIs, but a set of business process modeling and application development tools based on the Eclipse open source development platform.

Prospects

Telephony web services technology is in place today. What’s lacking is the widespread low cost access that web services from Google or Amazon offer. The problem is business models — PSTN calls still involve per-minute charges. But that won’t last since costs are falling and other funding models will soon appear. Then we can expect telephony to join the world of web services mashups, with an accompanying burst of services innovation.

Brough Turner may be reached by email or at +1 508 271 1312. Be sure to check out Brough’s blog at blogs.nmss.com/communications.

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Connecting PSTN Callers to IP Services


By Rich Pellegrini, Product Marketing, NMS Communications

IP-based telephony is beginning to deliver on its promise of multimedia enhanced services and the benefits of communications convergence. Soon, enterprises will be able to implement a converged SIP-based network, with IP voice and data sharing a common enterprise network with direct SIP trunking to the local communications carrier.

However, as IP-based telephony networks are being deployed, the reality is that the number of IP installations is miniscule compared to the installed base of PSTN circuit switches using SS7 signaling. As a result, the majority of subscribers and terminals that the SIP-based services need to reach are circuit-switched users. In order to reach this vast pool of wireline and wireless circuit-based users, IP telephony and enhanced SIP service networks must be connected to the PSTN and SS7 networks via gateway equipment. With the connection to and integration with the PSTN, SIP-based networks will be able to utilize and even enhance existing SS7 enabled applications such as text messaging, voicemail, VPN, 800 number services, intelligent call forwarding and pre-paid billing.

Solution Requirements

The media gateways required to connect IP-based solutions and enterprise branch offices and legacy PBXs must meet a number of platform, functional, interface, and availability requirements.

Platform

  • Turnkey, scalable solution
  • Proven technology
  • Interoperability with multiple switch vendors
  • Worldwide deployment capability

Functional

  • Intuitive web-based interface parameter configuration, routing table configuration, and system management information
  • Routing control for proper routing of calls offered on the PSTN or IP interfaces
  • Tracking of trunk resource availability, call state information, and call statistics
  • Software-based capacity upgrades (pay as you grow)

Interface

  • Connection to the PSTN via T1/E1 CAS, ISDN PRI (multiple variants), or SS7 ISUP (multiple ANSI and ITU variants)
  • Connection to IP infrastructure servers utilizing the SIP protocol

Availability

  • Redundant IP media interfaces
  • N+1 server architecture
  • SS7 interface redundancy

Providers of IP-based solutions require a simple and reliable interface to the PSTN to connect their solution to the majority of today’s subscribers. Multifunction gateway solutions that provide control, media, multiple signaling options, and routing in a scalable architecture simplify legacy network connectivity for solution providers, allowing them to concentrate on solution performance and functionality.

The Vision Solution

NMS’s Vision Media Gateway and complementary Signaling Server form a complete, scalable, highly avail­able turnkey solution for VoIP application developers who need to connect their IP-based service to legacy PSTN networks. On the VoIP side, the Vision solu­tion supports SIP signaling and a wide range of VoIP media. On the PSTN side, the Vision solution can be configured for SS7, ISDN, or CAS signaling. The Media Gateway provides call control and call routing functions, with routing configured using simple tables or under JavaScript control.

The Vision solution supports all the technical requirements listed in the previous section, and provides the following additional feature details:

  • Redundant Signaling Server configuration with call state checkpointing
  • A centralized Signaling Server (or redundant pair) that provides SS7 signaling for up to 20 Media Gateways
  • Dual Gigabit Ethernet for VoIP payload
  • Advanced call transfer features

In addition, the Vision solution can satisfy the require­ments for both enterprise and service provider markets. Figure 1 shows an example IP-based solution for the enterprise market.

Figure 1: Example IP-Based Enterprise Solution

Figure 2 illustrates an IP-based call center solution for the service provider market.

The Vision Media Gateway interfaces to the PSTN, providing access to the majority of the network callers. The PSTN offers multiple interface options with varying degrees of perfor­mance and functionality. ISDN PRI interfaces are usually suitable and very cost-effective for solutions that require limited trunking. As solutions start to scale up, as for large IP-based call centers, SS7 becomes more economical. A single SS7 link can handle the signaling traffic for thousands of voice circuits. Using SS7, however, provides not only a cost-saving solution, but also additional features and capabilities.

Figure 2: Example Service Provider IP-Based Call Center Solution


The Vision Difference

Feature

Benefit to Provider

Gateway functions to connect PSTN callers to IP-based resources

Expands the reach of IP enhanced services to PSTN callers

Allows applications to use SS7 signaling vs. ISDN PRI or CAS trunks

SS7 inter-machine trunks can be 40%–70% less expensive than ISDN PRI or CAS trunks for high density applications

High availability

Support for redundant, fault-tolerant configurations ensures continuous operation; NEBS chassis provides higher chassis availability

Proven performance/network interoperability

Leverages NMS’s CG and TX boards and our expertise in ISDN and SS7 signaling, allowing service providers to interface to virtually any PSTN worldwide

Rich Pellegrini may be reached by email or at +1 508 271 1293.

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Product News

New Low-Cost Signal Entry Panels (SEPs) Now Available


Beginning April 1, NMS is offering four new low-cost SEPs for creating RJ-48C (T1/E1) terminations for Open Access boards with either RJ-21 or MRJ-21 connectors. Refer to Table 1.

Part No.

Description

82923-1

SEP/cable kit – 1X (NMS RJ-21 to (8) RJ-48C interface)

82923-2

SEP/cable kit – 2X (NMS RJ-21 to (8) RJ-48C interface)

82924-1

SEP/cable kit – 1X (NMS MRJ-21 to (8) RJ-48C interface)

82924-2

SEP/cable kit – 2X (NMS MRJ-21 to (8) RJ-48C interface)

Table 1: New Signal Entry Panels

Each of the new SEPs has 2 RJ-21 connectors that connect to 16 RJ-48C connectors (see Figure 1). The difference between the SEP kits is the type and number of cables that are included:

  • 82923-1 includes one 6000-34231 cable (6-foot, shielded, RoHS, RJ-21 to RJ-21)
  • 82923-2 includes two 6000-34231 cables (6-foot, shielded, RoHS, RJ-21 to RJ-21)
  • 82924-1 includes one 6000-34236 cable (6-foot, shielded, RoHS, Mini RJ-21 to RJ-21)
  • 82924-2 includes two 6000-34236 cables (6-foot, shielded, RoHS, Mini RJ-21 to RJ-21)

Figure 1: New Low-Cost SEP Converts RJ-21 to RJ-48C Terminations

The 6000-34231 cable is a new cable for connecting SEPs to NMS CompactPCI boards that terminate with RJ-21 connectors on the RTM. This includes the TX 4000C, CG 6500C, CG 6565C, and two 8 T/E models of the CG 6060C (82374 and 82375). Customers who purchase a single-cable RJ-21 SEP (82782-1) connecting 8 RJ-48C connectors can later support 16 RJ-48C connectors on the same SEP by purchasing a second 6000-34231 cable.

The 6000-34236 cable is used to connect the mini RJ-21 connectors on the CG 6565 8T/E PCI boards to the 82924-1 and 82924-2 SEP panels. These boards are very popular replacements for two CG 6000 boards. Customers who buy a SEP kit for a single CG 6565 8T/E board can later support a second board by purchasing a second cable.

Note that these SEPs are equivalent to some existing SEPs but at much lower pricing. Technical information on these SEPs can be found on the NMS web site within the Software and Documentation downloads section. Contact your NMS sales representative for pricing information.

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RHEL ES 4.0 Support for SS7 Software v4.3


Red Hat Enterprise Linux (RHEL) ES 4.0 has been qualified for NMS’s SS7 Software v4.3. This software is now available for download from the Support and Downloads section of the NMS web site.

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CX 2000C-32-R to become RoHS-Compliant in April


The CX 2000C-32-R high-density analog station board (80654) will be the last NMS board to become RoHS-compliant when it is replaced by part number 82864 in April (refer to Table 2). The 82864 will feature a new RoHS-compliant power supply. As on the 80654, this power supply is on the RTM and can provide ring voltage for up to 32 telephones (12 ringing simultaneously). The new RoHS-compliant CX 2000C-32-R will be available on April 13, 2007, at which time NMS will discontinue the current nonRoHS-compliant board (80654).

Existing Model

Replacement Model

Part No. 80654

CX 2000C-32-R high-density analog station board with ringer

Part No. 80864

CX 2000C-32-R high-density analog station board with ringer, RoHS compliant

Table 2: Information on RoHS-Compliant CG 2000C-32-R

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China RoHS Information Available


Information that will help our Open Access and Vision customers who are pursuing RoHS compliance in China has been added to our NMS web site.

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Customer News

Ontario Systems Uses NMS’ Vision VoiceXML Server to Dramatically Increase Account Representative Productivity


Ontario Systems, the market leader in collection software solutions, is using NMS’s Vision VoiceXML server to provide innovative and powerful features in its Verified Contacts right-party contact system to make its customers even more productive.

“Ontario Systems’ collection software is key to the productivity of the more than 55,000 account representatives at the top healthcare and financial services companies in the country who use our collection software every day,” said Doug Allgood, Ontario Systems’ vice president of research and development. “A leader in its field, NMS has been a trusted partner for enabling technology for many years. We’ve come to count on the high performance, scalability and reliability of its products and innovations like the Vision servers that allow us to enhance our offerings with features that deliver more value to our customers.”

“Ontario Systems is a leader in the collections software industry, with solutions like the Verified Contacts system that help their customers improve productivity and increase profits,” said John Orlando, chief marketing officer for NMS. “This is just one example of the types of high performance interactive solutions that are made possible using the Vision VoiceXML server.”

About Ontario Systems

Ontario Systems, LLC provides debt collection software to organizations that manage large volumes of accounts receivables. These include hospitals, collection agencies, collection attorneys and bankcard issuers. As the largest provider of receivables management products, including the Artiva, FACS and CT Vision systems, Ontario Systems is recognized throughout the industry as the leader in technology and customer service. With operations in Indiana and Ohio, it is a privately held company headquartered in Muncie, Ind. For more information, visit www.ontariosystems.com.

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Support Tip

Measuring Echo Delay on the Network


All NMS AG and CG boards support DSP-based echo cancellers. The DSP-based echo canceller can be configured to use a filter length between 2 ms and 64 ms. The higher the filter length used, the more DSP MIPS are consumed in the echo cancellation operation. Therefore, in order to make optimized uses of available DSP resources, it is recommended that application developers measure the echo on the network so that the minimum necessary filter length is used.

The CG 6565(C) and CG 6060(C) employ a new approach to echo cancellation. A ASIC-based echo canceller is available on all trunk lines, so that each input DS0 channel can have G.168-compliant echo cancellation with 64 ms tails. This is a significant improvement over earlier CG boards, not only because of improved algorithm quality and tail length, but because the DSP chips are no longer required to perform echo cancellation, allowing that DSP capacity to be used for other purposes. If there are echo problems when using this ASIC-based echo canceller, it is important to understand if the echo on the network is beyond 64ms.

To measure the echo delay (the roundtrip between outgoing and incoming echo signal) on a network, use the DSP-based echo canceller. The API adiModifyEchoCanceller() can be used to query the current status of the echo canceller by setting the bit ADI_ECHOCANCEL_STATUS in the mode field of the ADI_ECHOCANCEL_PARMS structure. Setting the bit ADI_ECHOCANCEL_AUTOSTATUS instead will cause the echo canceller to send an event to the application each time the status of the echo canceller changes.

The echo canceller will generate ADIEVN_ECHOCANCEL_STATUS events. The status information is stored in the event buffer and arranged according to the ADI_ECHOCANCEL_STATUS_INFO structure.

typedef struct
  {
  WORD status;   /* Echo canceller status flags  */
  WORD ERL;      /* Echo Return Loss             */
  WORD ERLE;     /* Echo Return Loss Enhancement */
  WORD sndLevel; /* Level of the sent signal     */
  WORD rcvLevel; /* Level of the received signal */
  WORD refPoint; /* Reflection point location    */
} ADI_ECHOCANCEL_STATUS_INFO;

The field refPoint in this structure contains the position of the maximum value in the H register in 8 kHz sample increments.

For example if the returned value of refPoint is 120, the reflection point is 120 x 125 µs = 15 ms, which means that a minimum tail length of 20 ms is required for DSP-based echo cancellers, while the echo should be cancelled for ASIC-based echo cancellers on CG6565(C) and CG6060(C) boards.

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For additional support tips, visit the Technical Notes section of the NMS web site.

Spotlight Events

Join us at NGN+S Spring Conference in Tokyo, May 29, 2007


About the show

For the past three years, Ric Telecom’s NGN+S Spring is the only event on NGN in Japan, drawing in telecommunications carriers, ASPs, and ISPs. The conference presents a clearer vision for the future of NGN network and NGN services. This year, the theme is “What NGN services are in demand”, focusing on NGN services.

NMS will be partnering up with Star Communications in a joint booth and showcasing our MG 7000A AdvancedTCA media processing blade. If you are interested in arranging a meeting with a NMS representative at the show, please email saya_miwa@nmss.com, indicating your preferred meeting date and area of interest.

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Web Seminar


 On April 24, 2007,
NMS will present the web seminar

“ATCA vs. MicroTCA: Options for Developing Highly Available Telecom Solutions”

Application developers and network equipment providers seeking to provide five-nines solutions to their service provider customers have a choice of architectures — AdvancedTCA and the emerging MicroTCA. While both architectures can meet carrier-grade telecom equipment requirements, each has advantages for different applications. This webinar describes the similarities and differences of these two form factors, and discusses the markets for which each is best suited.

Who Should Attend

Application developers, network equipment providers, product managers, and systems architects developing and deploying highly available applications for fixed and mobile telecom environments.

Register now for this web seminar on April 24 at 11:00 am Eastern / 8:00 am Pacific.

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Save the Date! Connect 2007 Event Dates Set for Americas, Europe, and Asia


Connect 2007, hosted by NMS, is a series of global conferences that target key business and technical decision makers who are interested in learning about the vital drivers shaping the communications ecosystem — the business issues, the market drivers, the technology challenges, the compelling applications, and the end user demands. This year events take place in Boston (USA), Guilin (China) and Madrid (Spain).

Connect Americas Conference
October 2–3 Boston, MA, USA

Connect Asia Conference
October 17–18 Guilin, China

Connect EMEA Conference
November 7–8 Madrid, Spain

We look forward to making connections with customers, prospects, and industry leaders at future Connect events. To sign-up for event updates, please click here to fill out a brief form.

If you are interested in speaking or exhibiting, please email Christine Krajewski or call +1 508 271 1129.

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For detailed information on NMS’s participation in industry events, conferences, and seminars, visit the Events Calendar.

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