Tag Archives: QoS

Telco Systems’ Access211 ATA with QoS Chosen for SIP VoIP

January 25, 2005


The Access211 ATA (Analog Telephone Adapter) delivers broadband Internet telephony and faxing over standard analog gear, and includes a built-in router, lifeline POTS support, and prioritization of voice over data for QoS.

Telco Systems announced the selection of its Access211 VoIP ATA as the basis of a SIP-based VoIP calling service by GlobalTouch Telecom, Inc, which has initiated a multi-year, multi-million dollar contract for the Access211 with Telco Systems.

Called SIPTalk, the service allows subscribers to use standard analog phones and fax machines for broadband telephony and Internet T.38 faxing, while prioritizing voice over data to deliver higher voice quality through traffic shaping, a technology the company asserts is not available on any other consumer VoIP ATA.

It also incorporates lifeline POTS support, which retains analog phone line access to E911 services in the event of power or Internet outages. Other features of the Access211 include the integrated router for simplicity and reduced desktop clutter, custom calling features such as distinctive ringing, and two telephone ports that concurrently support compression for two different phone numbers.

GlobalTouch’s SIPTalk product line starts with a new $9.95 per month residential plan with 200 minutes of calling anywhere in the US and Canada. Residential and small business plans start at $14.95 per month, rising to unlimited US and Canada use for $19.95 per month. All SIPTalk plans include low international rates.

Telchemy Software Embedded in Pivetal’s New VoIP Speech Quality Monitors

December 6, 2004


telchemyPivetal will provide real-time VoIP-specific monitoring of user-perceived quality, thanks to Telchemy’s call quality analysis software.

Telchemy, Inc and Pivetal, Ltd announced today that Pivetal has embedded Telchemy’s VQmon/SA Stream Analysis call quality algorithms in Pivetal’s new QspeeQ family of VoIP Speech Quality Monitoring products, allowing Pivetal customers to assesses user-perceived VoIP quality in real time.

This new VoIP-specific capability further enhances the ability of Pivetal’s Cortex Service Assurance Suite to identify and fix network issues before they impact on service quality levels. This meets the needs of service providers, large and small, to monitor and enforce SLAs (Service Level Agreements) at the customer premise, as they begin rolling out real-time multimedia services such as VoIP.

Traditional call quality management systems have not so far provided the diagnostic tools needed to support the SLAs for new IP-based services, generating a demand for more sophisticated capabilities.

Telchemy’s VQmon provides real-time analysis of VoIP calls, generating quality metrics including listening and conversational quality scores, and detailed diagnostic information on the severity and distribution of packet loss, discards and jitter. VQmon is standards-based call quality monitoring software that supports the IETF RFC 3611 (RTCP XR) Protocol and the new QoS reporting protocols for ITU’s H.323, H.248 and G.799.1 standards.

Telco Systems Partners with Phillips to Sell Carrier-Class IP Systems to ILECs

November 26, 2004


telcosystemsTelco Systems announced a distribution agreement with Phillips Communications & Equipment Company to deliver carrier-class copper and fiber-based transport and access equipment to the ILEC market.

Telco Systems, Inc is securing an early position in the move by ILECs (Incumbent Local Exchange Carriers) to deliver fiber services directly to end user customers by signing an agreement with Phillips Communications & Equipment Co, which will distribute Telco Systems’ end-to-end active Ethernet FTTx (Fiber-To-The-Home/Curb/Business) family of carrier-class IP aggregation and core routing switching platforms and VoIP media gateways nationwide.

Phillips will be offering Telco Systems’ VoIP, fiber-to-the-home/business and multi-service solutions, comprising its Access211 and 201, EdgeGate CPE and IP Switch platforms for VoIP and FTTx; as well as its multi-service EdgeLink OTM1000, EdgeLink Hub, and EdgeLink100 family of products which provide DS1, DS3, and Ethernet service delivery.

The Telco Systems FTTx end-to-end solution enables a service provider to connect end-users over fiber from their home or office throughout a community, metro area, campus environment, multi-dwelling unit or multi-tenant unit to an IP backbone and deliver VoIP Internet voice service, video, data and broadband Internet access. The company’s Access211 and Access 201 VoIP gateway connects to either a DSL or cable modem to enable service providers to offer VoIP service over a high-speed Internet connection.

Phillips Communications & Equipment Co was established in 1979 to decommission central offices and to restore and resell the equipment, but has since transitioned to become a distributor of new, and used central office and transmission equipment to the ILECs.

Telco Systems, a wholly owned subsidiary of BATM Advanced Communications (London stock exchange ticker symbol: BVC) focuses on integrating transport, access, and packet technologies onto carrier-class IP packet-based platforms, such as high-capacity, self-routing switches with extensive QoS (Quality of Service).

MERA Unveils High Capacity Session Controller

September 22, 2004


Fully Distributed Session Controller Architecture Provides Major VoIP Carriers with Greater Flexibility, Resiliency, and Scalability of Networks and Services.

Toronto, ON, Canada – September 20, 2004 – MERA Systems, Inc., a supplier of full-featured, reliable and scalable session controllers for carriers of any size, today announced the availability of its new fully distributed session controller architecture. The new architecture has been purpose-built for large telcos such as Tier 1, major national and international carriers and delivers enhanced scalability, reliability and simplified management of VoIP networks.

Distributed architecture: what’s the difference?
In today’s mature VoIP world, physical separation of the session controller components is becoming a critical factor, ever more so for carriers that sit on the top of the telecom food chain and operate complex networks based on multivendor technologies.

MERA provides a fully distributed session controller solution that delivers this “five-nines” reliability along with other benefits critical for a top-tier service provider. The MERA MVTS session controller consists of several interconnected signaling and media servers. The signaling servers share a single routing table and can therefore efficiently manage VoIP traffic throughout the entire network, irrespective of their physical location.

MERA’s distributed architecture makes a big difference for big carriers:

* Increased capacity, easy scalability and reduced CAPEX: the clustered architecture of MERA’s session controller allows a capacity increase up to 40, 000 concurrent calls. The solution easily scales by adding extra servers, and is significantly cheaper than several single-server session controllers put together for the same capacity.

* Added reliability: Along with regular backup redundancy, MERA’s distributed session controller is “redundant in itself”. Every signaling server is backed up by all the other signaling severs, and the same goes for media servers, which provides a new, unprecedented level of reliability to the carrier’s network with downtime next to none.

* Intelligent traffic routing and QoS enforcement: as all signaling servers in a MERA cluster share a common database on routes and policies, they efficiently distribute calls throughout the entire network irrespective of the geographical location of each server. Furthermore, MERA’s session controller delivers comprehensive, end-to-end signaling capabilities. MERA’s signaling servers perform real-time bandwidth mapping and allocation, and send media flows across channels with better bandwidth, thus providing for greater service availability and customer-centric QoS policies.

* Simplified configuration and management: The MERA Session Controller makes Tier 1 networks easily manageable despite their size and complexity. All components in a MERA cluster share a single set of configuration files, with all updates automatically synchronized between the servers. It means carriers only need to configure the system once, while a collection of separate boxes would require configuring each one independently, with inevitable delays in service rollout and the need for extra resources to administrate geographically dispersed network elements. In fact, MERA’s configuration is so easy that it can be performed by the carrier’s staff, unlike many other solutions that require on-site presence of the vendor to configure the system.

Customer Deployment
Infotel Communication S.p.A., an Italy-based VoIP carrier, has deployed the MERA MVTS session controller with distributed architecture in its nationwide VoIP network for delivery of real-time voice services in Italy and abroad. MERA’s cluster at Infotel Communication is composed of the MVTS signaling server and two media servers deployed in Milan, and handles VoIP traffic from 12 POPs in Italy and Europe, over 20 Tier 1 and Tier 2 international carriers, and a network of resellers and phone shops. All signaling flows are concentrated by the MVTS signaling server, which distributes calls between the media servers. With this architecture, media information does not have to follow the signaling flows, thus offloading the signaling server for higher traffic throughput.

“MERA’s distributed session controller is a highly efficient tool for providing high-quality termination, origination and wholesale services to our customers throughout the country”, says Luigi Ghirardi, Technical Director at Infotel Communication.” The convenient architecture of MERA enables us to keep our network simple and easily manageable, ensuring fast and easy rollout of new services. With MERA’s solution we can grow our network in no time: adding an extra server to the current configuration will take less than five minutes, increasing capacity by 1,500 concurrent calls. Along with its feature-rich functionality and ready scalability, MERA’s MVTS has saved us a lot of money: for instance, deployment of back-to-back gateways would have cost us from 100% to 200% depending on the functionalities needed, and the management overhead would mean a further increase of approximately 30% per year. Furthermore, a distributed and “clustered” solution is even more cost-effective – it’s the economy of scale that is at work here. The bigger your capacity, the more you gain, which is not always the case with multiple single-box units.”

PointOne Managed IP Readies VoIP

July 15, 2004


StarPoint IP™ Provides More Reliable Phone Service than Non-Facilities Based Alternatives; Cable Operators Provide Local Presence

AUSTIN, July 14, 2004 — PointOne today announced details of its StarPoint IP™ service, which is bringing superior residential and SOHO Internet Voice services over the largest managed Voice over IP network in North America. PointOne is teaming with regional cable operators to provide local facilities-based VoIP services to consumers accustomed to reliable and high-quality phone services.

The service, available as StarPoint IP™ Home and StarPoint IP™ Office, will be delivered through cable operators to ensure the highest Quality of Service (QoS) and provide consumers with a local presence.

PointOne has invested over $100 million in developing a facilities-based network that covers 75% of the U.S. population with a privately managed IP network, which ensures end-to-end network performance rivaling the public switched network. PointOne’s converged voice and data IP network will in most cases carry the entire phone call without touching the public Internet, ensuring that it can constantly maintain quality and performance.

“Broadband communications has evolved to the point where VoIP is ready for prime time, so long as service providers deliver the same quality and reliability that consumers have come to expect from the public phone system,” said Mike Holloway, president and CEO of PointOne. “With PointOne’s network behind it, StarPoint IP™ will become the platform for new voice and enhanced applications that are only possible over an IP-based network.”

The ubiquity of the Internet makes it possible for entrepreneurs to offer hosted VoIP services with no investment in networking and operations infrastructure. But such services, which are being announced on an almost weekly basis, do not have the ability to guarantee – or even monitor — consistent quality and reliability. Because local broadband suppliers are not involved in providing the service, such services likely dilute network performance in local areas as subscribers increase.

PointOne is teaming with cable operators throughout the U.S. to provide superior IP phone service to their customers, along with locally set rates, service and installation.“StarPoint IP provides a complete turnkey solution for cable providers,” said Sam Shiffman Executive Vice President of PointOne. “We provide a server at the cable head end that connects the cable provider’s network to the PointOne network to ensure a managed connection and assured bandwidth. This delivers the highest quality of service currently available in the industry.”

StarPoint IP™ will deliver a full range of robust features to residential and SOHO customers, including long distance; local service with e911, directory assistance and operator services; enhanced features including call waiting, call return, call forward, caller ID/ID block; voice mail; 3 – way calling, conferencing; and a service provider web portal.

In addition, StarPoint IP™ Office will include abbreviated dialing; call hunt; reservation-less and multi-way conferencing; an Auto Attendant to receive incoming calls via an automated voice response system that provides directory listing and automatic call forwarding based on user extension; and free fax line.

Telchemy releases new VQMON/EP Release 2.1

June 8, 2004


VQmon/EP 2.1 Software Supports RFC3611 and QoS Reporting Requirements for G.799.1, H.323 and Megaco

SUWANEE, Georgia –June 7, 2004 – Telchemy (R) Incorporated, the global leader in VoIP performance management technology, today announced VQmon (R) /EP (End Point) Release 2.1 which provides integrated call quality monitoring and reporting for IP phones and media/trunking gateways. As more service providers and enterprises deploy VoIP, real-time performance monitoring becomes critical for reliable service operation. VQmon/EP 2.1 supports RFC3611 (RTCP XR) and QoS reporting requirements for IP phones and gateways — including the new G.799.1 trunking gateway standard.

Other key features of VQmon/EP Release 2.1 include:

• full extended E model (G.107 with TS 101 329-5 Annex E)

• real-time alert generation

• support for jitter buffers that discard fractional packets

• dynamic configuration of VQmon-based probes, including Telchemy’s new SQprobe (TM).

VQmon/EP 2.1 will be available July 31, 2004.

About VQmon/EP

A member of Telchemy’s VQmon family of software performance management products, VQmon/EP is used in IP endpoints, such as IP phones and media/trunking gateways, to monitor the quality of live calls and provide the call quality and diagnostic metrics needed to support management protocols and problem diagnosis. VQmon/EP supports real-time thresholding, generates and interprets RTCP XR payloads and provides the metrics for signaling-based QoS reports. The software is fast, efficient, scalable and highly portable which minimizes costly implementation time and reduces time to market. VQmon is fully compatible with VQmon(R) /SA (Stream Analysis) and a wide range of VoIP test equipment.

Prominence Announces New Management Solution

May 5, 2004


Prominence Networks announces new configuration and management solution for IP telephony and videoconferencing over diffServ networks. MediaIP Release 3.0 incorporates diffServ capability, redundant configuration & support for distributed, multi-site VoIP deployments.

Holmdel, New Jersey, May 4, 2004 – Enhancing and extending the configuration and management of Differentiated Class of Service (DiffServ) networks, Prominence Networks, Inc. today announced MediaIP Release 3.0, the latest version of its MediaIP service control and management solution for IP telephony and videoconferencing. In addition, MediaIP Release 3.0’s advanced redundancy and clustering features support large, distributed, multi-site IP Telephony deployments requiring increased reliability and availability.

Many enterprises have adopted the DiffServ quality of service (QoS) model to transport real-time traffic such as voice and videoconferencing over their converged IP networks. MediaIP Release 3.0 extends DiffServ’s packet marking, classification and policing mechanisms with:

– End-to-end network intelligence to ensure bandwidth is available end-to-end for high quality IP voice and video calls.

– Network wide call admissions control to admit calls only if sufficient bandwidth is available throughout the network.

– Simplified DiffServ configuration and management through automation and dynamic bandwidth management.

Availability and Reliability for Enterprise Deployments

In addition, as enterprises deploy IP telephony across their multi-site footprints, they require high availability and reliability. MediaIP Release 3.0 meets these needs through a new active/standby cluster configuration with 1+1 physical redundancy to provide higher levels of availability. Moreover, MediaIP Release 3.0 has been engineered to support multiple gateways, gatekeeper zones and IP PBX clustering to meet the needs of large enterprise deployments that feature distributed architectures for redundancy, scalability and resiliency.

“Configuring, managing and maintaining real-time applications such as VoIP and IP videoconferencing with DiffServ can be a challenge as the network evolves and traffic patterns change,” notes Sid Nag, Prominence Networks Founder and CEO.

“MediaIP Release 3.0’s automated configuration and management simplifies the configuration process, and its dynamic bandwidth management adapts to real-time traffic changes automatically. MediaIP’s end-to-end network view enhances DiffServ’s capabilities to support network wide call admissions control; calls are only admitted if there is sufficient bandwidth at every hop.”

Key Features/Capabilities of MediaIP Release 3.0:

Differentiated Class of Service Support

– End-to-end network view supporting dynamic bandwidth management

– Enhanced, network-wide call admissions control

– Automated configuration and management of router queues and policies

Redundant Configuration for Enhanced Availability and Reliability

– Active/Standby cluster configuration with 1+1 physical redundancy

– Cluster configuration for the MediaIP Controller and Director

– Database replication and backup; automatic failover

– Support for Distributed, Multi-Site IP Telephony Deployments

– Support for redundant IP PBX configurations

– Multiple gatekeeper zones support

– Multiple gateway support

Pricing and Availability

Prominence Networks Release 3.0 products are available immediately. The Prominence MediaIP Service Control Solution is flexibly priced to accommodate small branch offices to Fortune 50 corporate headquarters. Prominence’s MediaIP Service Controller is available in appliance and 1U server models and is licensed based on the number of simultaneous calls supported. The standard, rack mountable MediaIP Director is licensed by the number of sites managed.

Ingate launches partner program at Spring VON

March 15, 2004


Extends Successful SIP Sales Support, Certification and Training Program – Already Established Overseas – to U.S. Market.


HOLLIS, N.H., March 15, 2004 – Ingate® Systems, which produces and sells the world’s only fully Session Initiation Protocol (SIP)-capable enterprise firewalls, is launching the Ingate Partner Program, an extensive sales support, SIP certification and training program for resellers of telecommunications products, in the United States. The Partner Program is already a success in Europe, and is designed to create a powerful network of SIP-educated resellers that are well positioned to meet the demands of businesses looking to integrate VoIP or any SIP-based realtime communications applications into their enterprise.

The Ingate Partner Program will be announced at Spring VON 2004 on March 29.

The Ingate Partner Program brings together a broad knowledge base on the implementation and enterprise integration of realtime communications applications based on SIP. The standard protocol for VoIP, instant messaging, presence and a host of cost-saving business tools, SIP is quickly becoming a necessity for companies of all sizes looking to remain competitive. This growing demand for SIP-enabling products that can be integrated quickly, effectively and on-budget has many resellers eager for a resource from which they can draw information and sales support.

Key components of the program include education and training opportunities to prepare resellers for SIP-focused sales. Certification training will be available in the following areas: Ingate Firewalls, SIP, VPN and QoS, and sales. Becoming an Ingate Certified SIP Partner assures customers that resellers are fully knowledgeable about SIP and its enterprise implementation, and can deliver informed technical support. “NAT traversal and security are still issues facing IT managers as they SIP-enable their enterprises, but there are solutions that make sense for enterprises of all sizes,” said Olle Westerberg, Chief Executive Officer, Ingate Systems. “Resellers must be able to understand and anticipate questions about integrating SIP and be prepared with the right products for their customers.”

“The Ingate Partner Program started in 2003 in Stockholm. SIP adoption was starting to really take off, and customers were looking to adapt their networks. We found that resellers were eager to tap into our extensive knowledge of SIP and its implementation. Today, Ingate’s Partner Program in Sweden boasts 25 members and is still growing strong,” said Westerberg. “We look forward to meeting the needs of our U.S.-based resellers as well.”

There is no fee to join the Ingate Partner Program.

Partners will receive substantial sales support through Ingate partner WEB, a repository of technical information, sales materials, template sales presentations and a product roadmap for enterprise-wide SIP integration. Information about Ingate activities will also be available through the Ingate partner WEB.

Ingate Partners will also enjoy the following benefits:

• An Ingate Firewall 1200 demo or Ingate Firewall 1400 demo unit at a reduced price.

• E-mail newsletter

• Public relations and marketing support

• Posting of logo on the Ingate home page (www.ingate.com)

• Active sales support