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September 29, 2005

VoIP in the banking sector

Banks and financial institutions, particularly small and mid-sized ones have been quick to appreciate the benefits that VoIP can offer in terms of cost savings and enhancing customer experience. Other benefits include more robust disaster recovery systems, improved CRM, greater mobility for the staff, etc. According to Susan Cournover, analyst with Gartner Inc., VoIP will be indispensable in 70% of the businesses by 2008.

Currently, the education sector is the most enthusiastic adopter of VoIP and accounts for around 35% of the total market, healthcare and financial institutions account for 16% and less than 16% of the market, respectively. Small and mid-sized banks have been the early adopters of VoIP and are using it to improve the service of their customer contact points.

Banks have been a trifle slow in adopting VoIP as there is more at stake with banking services as compared to other businesses. However, the attraction of almost 50% savings in communication bills and the maturing of VoIP have meant that financial institutions are venturing forth to adopt VoIP.

SouthTrust Bank, based in Birmingham, Alabama, saves in excess of $ 5 million per annum after having installed a VoIP solution by Cisco Systems. The savings have accrued due to cost reductions across the communications setup. These include a 20% reduction in cost related to frame-relay circuits, a 51.3% reduction in moving and adding costs, a 94% reduction in conference call costs, etc. The bank is also targeting a 15.4% reduction in the voice and data systems maintenance costs. For the bank, a major advantage of implementing VoIP is that the safety and integrity of its data will not be compromised in the event of a mishap at any of its locations.

SIP security

SIP provides security against activities that are aimed at disrupting aspects of SIP operation. This includes activities such as eavesdropping and hijacking a call. However, SIP cannot prevent a DoS attack initiated by flooding the SIP server. Hardened servers and up to date firewalls are the best defense against DoS attacks.

SIP can ensure media security such that it would be well nigh impossible for attackers to decipher data. SIP uses Secure RTP for the encryption and authentication of every single media packet. However, if VoIP devices running on the SIP protocol communicate over a LAN such as an Ethernet hub, they become vulnerable to sniffing attacks. Enterprises prefer switched Ethernet over hubs as they are more resistant to sniffing attacks.

However, a tap can be inserted into the switch by the attackers in order to access data. In order to prevent the spoofing of IP addresses, ingress source filtering is done. Packets with suspicious IP addresses can be detected by this technique. SIP security mechanisms prevent attackers using spoofed IP addresses to fake a caller ID. The issue of spam in VoIP is only partially controlled by traditional techniques such as black / white listing; content analysis is not possible with VoIP. Even though it is possible to authenticate a call with SIP, it is not adequate protection against spam.

Security at the design stage

According to security experts, it is important that VoIP vendors and customers factor in the security aspects along with cost and performance when considering a VoIP network. Chris Thatcher, national practice leader, Dimension Data Holdings is of the opinion that the design of VoIP systems has not covered the security aspects satisfactorily. VoIP networks are exposed to risks such as distributed denial-of-service attacks, spoofing, worms, viruses, Trojans, etc. Vbombing is a threat unique to VoIP networks in which a VoIP console is bombarded with voice mails and can crash.

Since the market is relatively new, awareness regarding the security threats to VoIP networks is not very high and on occasions even the vendors are not very keen to discuss the fallibility of their applications. Scripts for launching attacks on VoIP networks can be found on several hack sites. Transport-layer security can be handled by firewalls but many attacks target the application layer, which in case of several VoIP applications, is based on SIP. SIP is not unlike SMTP and HTTP and all the security issues that are present in emails are a threat to VoIP as well.

VoIP and gaming

VoIP is poised to give gaming companies an advantage in the battle for IP telephony supremacy. Microsoft and Sony Corp. offer gaming networks, namely xBox Live and Playstation, which are VoIP enabled. eweek.com reports:

"It's tough to say what gaming consoles or services will offer in terms of communications capabilities, but there's no reason to think that additional [VOIP] platforms won't grow out of those systems," said Joe Laszlo, analyst with New York-based Jupiter Research.

Read More: Playing Games with VOIP

September 28, 2005

eBay and Skype

Meg Whitman, CEO, eBay, states that eBay's acquisition of Skype will help it to leverage Skype’s technology to further the growth of PayPal and eBay. However, according to some business analysts, the manner in which eBay and PayPal operate does not justify the purchase of Skype. This is because VoIP will probably not add much to the already established online trading system that eBay has perfected.

Also, eBay and PayPal target the consumer to consumer and business to consumer market; if the company wishes to move into business to business as well then Skype may not be the right choice as Skype too does not spring readily to mind when discussing business to business VoIP. Moreover, even though eBay has acquired Skype and its base of over 50 million users, it will still have to face serious competition from Google and Microsoft.

Google has introduced Google Talk Beta, which has the advantage of running on the open SIP and will allow Google greater interoperability. Google may team up with Earthlink and SIPphone. LCS 2005 is Microsoft’s venture into providing VoIP services to businesses. Microsoft has also purchased Teleo, which is a VoIP provider. Yahoo was already providing PC to PC and PSTN call facility to the users of its IM, its purchase of Dialpad Communications will allow it to add VoIP calls facility to its IM service.

One advantageous manner in which eBay can leverage its purchase of Skype is to employ its knowledge of the preferences of its customers and target them accordingly using VoIP.

VoIP vulnerabilities

The use of PCs in conducting VoIP operations makes VoIP systems susceptible to a host of threats ranging from spam and spoofing to worms and Trojans. One way of reducing threats to VoIP is to run voice and data networks separately. eweek.com reports:

Dr. Shashi Phoha, director of the Information Technology Laboratory at the National Institute of Standards and Technology, said she thinks that the growth of VOIP technology brings with it some significant risks that users need to be prepared to address.

Read More: 'Severe' Vulnerabilities Are Possible in VOIP, Official Warns

FCC regulation regarding 911 availability

The Federal Communications Commission (FCC) requires that VoIP providers intimate their customers about the unreliability of their VoIP service. eweek.com reports:

In a public notice Tuesday, the FCC said it will hold off enforcement of the rules until Oct. 31 if VOIP operators have acknowledgements from less than 90 percent of their customers.

Read More: Regulators Give Some VOIP Operators More Time

The telecommunications bill

The US House Committee on Energy and Commerce has drafted a 77-page bill that regulates DSL, cable modem, and other broadband services so that providers do not discriminate between access providers and content providers. eweek.com reports:

The hands-off approach is favored by the long-entrenched service providers, such as the RBOCs (Regional Bell Operating Companies), but it is not fully embraced by all users, particularly small businesses.

Read More:Small Businesses Wary of Telecommunications Bill

Open source vs. proprietary technology

The main reason for VoIP providers to push for open standards is to reduce their dependence on proprietary software, which sometimes inhibits their ability to provide quality service. eweek.com reports:

Silicon Valley is fast moving into the world of telephony, and it is dragging the contest between open and proprietary code along with it.

Read More: Telephony Battle: Open Vs. Proprietary

Hosted VoIP and enterprise-level service

Increasingly, companies are finding out that the services provided by their vendors are not enterprise-strength. There are around 400 VoIP service providers in the US but very few provide facilities to handle the call volumes of a company and services such as IP PBX support, broadband VoIP, etc.

Companies such as Qwest are taking into consideration the customer’s requirements and in October 2004, Qwest introduced IP Centrex Prime, which is an IP Centrex service for companies that have their offices distributed over several locations. Qwest also provides another service known as OneFlex, which enables small and medium businesses use VoIP phones for conference calls and managing functions on their own.

Similarly, the PremierSERV Hosted IP Communication Service provided by SBC Communications is aimed at enabling SMBs to customize the features as per their needs, this includes find me/follow me capabilities. Users can access calls over IP networks and PSTN. SMBs are obtaining their VoIP services from CLECs and regional providers.

The disadvantage with hosted VoIP providers is that they are dependent on the telephone loops and if these are down then the VoIP service is affected. The number of VoIP providers has increased due the low entry barriers in this trade. The level of service offered by the providers is affected by the quality of the instruments provided by the vendors.

According to Infonetics Research Inc., the growth in demand for voice application servers, soft switches, etc indicates that the providers are purchasing the type of equipment that will help them to provide better service. Covad provides services such as virtual PBX, call logs, visual voice mails, etc; SBC provides services such as hosted IP communication, unified messaging, plug and play, etc; Qwest provides services such as caller ID, call waiting, online dashboard, etc.

The growth of VoIP

By 2006, more than 66% of the Global 2000 companies will implement VoIP. At present there are close to 400 VoIP only providers in the US and even network vendors, including the RBOCs are now providing IP options. Cost reduction achieved by running voice traffic on packet-switched networks is the main reason for companies switching over to VoIP.

Savings on toll-charges, shifting, operating expenses are possible with VoIP. However, it must be borne in mind that the capital cost of IP applications is not very low, even if the VoIP network is hosted by a third party, it does not automatically translate into savings. New applications such as unified messaging and instant messaging can be implemented in a smooth manner by converging voice and data networks. This is a big reason for companies to migrate to IP-driven networks.

The advantages offered by the new applications and improvements in IP PBXs resulting in better security and QoS are an attraction for many companies. These developments, which keep cost in mind, are helping in bridging the disconnect between customer needs and vendor marketing, which had occurred initially because vendors approached the issue of VoIP adoption from a technology perspective.

To encourage companies to adopt VoIP without losing out on their investment in the existing systems, vendors are offering hybrid IP systems that allow a gradual convergence. Thus, companies need not wait for their existing equipment to come to the end of its life-cycle. In Illinois, the village of Lombard deployed an IP telephony network purchased from Cisco. The network consisted of Cisco switches, unity voice mail, dual Call managers and around 240 IP phones. The village has around 42,000 residents and has used the IP system to connect the Village Hall, the police station, fire station and the water works department. The village of Lombardy is an example of a community or organization that wants to do away with its existing system and is in a good position to employ VoIP from scratch.

In order to achieve a truly converged network, network monitoring and balancing of the traffic need to be managed. To this end, the enhanced remote monitoring diagnostics offered by Avaya should be of help. The migration to a VoIP-enabled network should happen on a large scale in the next few years because the existing networks installed around the year 2000 will be coming to the end of their lifecycle. The onus is on the IP- systems vendors to ensure that the transition for its customers is as smooth as possible.

Converged networks and converged threats

VoIP networks are susceptible to threats that originate at the data networks. Companies are well aware of and therefore prepared to handle the threats that originate in the VoIP networks but managing the threats that can occur due to the convergence of networks may well be a hidden cost that most companies need to factor.

The traditional tools of protecting data do not afford adequate protection to VoIP data. In fact, even modems that can be accessed via phone lines possess a threat to VoIP data. A hacker can use the company’s phone lines to make expensive long distance calls, misuse the bandwidth, and commit subscription frauds. A converged network brings together data and voice networks and in doing so, it exposes one network to threats from the other. A thorough consideration of the threats helps to deploy the necessary security tools, often without much additional expenditure.

SecureLogix Corp. is a company that offers VoIP monitoring products that manage the traffic that passes through voice gateway devices in a manner that is similar to how firewalls manage data traffic. Other security measures include boosting perimeter defenses, encrypting data, hardening servers, etc.

September 27, 2005

VoIP for telecommuters

In the last week of August 2005, 3Com Corporation and Ingate Systems came together to provide VoIP facility to telecommuters who can now hope to avail enterprise-level IP telephony, even when on the move.

The new IP Telecommuting Module that has been designed by 3Com enables remote employees to securely access corporate VoIP applications. Companies that are already using 3Com can benefit from the seamless convergence provided by the module. The module is based on open standards and uses SIP. The employees can use their laptops to make telephone calls from any place with an Internet connection, by using the corporate VoIP system.

Ingate System technologies will help 3Com to manage the Network Address Translation (NAT) aspects related to the SIP communication. A single module can handle up to 100 users simultaneously. The module is available for $80/user.

Open source VoIP products

Open source VoIP products include

• sipXpbx, which is an SIP PBX aimed at SMBs.

• sipXregistry, which is an SIP registrar and redirect server.

• sipXpulisher, which is a subscribe/notify server for SIP event subscriptions.

• sipXcal, which is a call processing library.

• sipXvxml, which is a voice processing engine. eweek.com reports:

The first open-source release under SIPfoundry will be Pingtel's source code for its SIPxchange IP PBX and Instant Xpressa soft phones, under the GNU Lesser General Public License.

Read More: Push on to Make VOIP Open Source

VoIP migration

When Provident Funding Associates LP decided to shift to VoIP to manage its growing communication needs in a cost-effective manner, it assessed VoIP solutions provided by a host of vendors including Cisco Systems Inc., Nortel Networks Ltd., Avaya Inc., etc.

The investments that were required in implementing solutions developed by the larger vendors did not assure the company of an early ROI. Also, the solutions proposed required the purchase and installation of around 200 discrete items, which would have made the management of the VoIP network almost as difficult as the legacy network that they wanted to phase out. VoIP systems require several pieces of equipment such as media servers, voice mail servers, gateways, etc that need to work together, whether housed together or separately.

Provident wanted a system that would integrate with solutions in-house and the company need not have to hire an outsider for managing the system. Large companies can promote their proprietary standards on the strength of their already established client base. Provident finally decided to implement Zultys MX250, which is based on open standards and required only seven components to run the VoIP system. The MX250 provides a PBX and Internet gateway in one box.

The ZultysMX25 can be employed as a SIP gateway and can also handle up to 30 simultaneous calls. Provident also benefited from the fact that any SIP-based handset can work with the Zultys equipment. Implementing the Zultys VoIP system has enabled the company to save on the cost of internal calling which used to be around $ 300,000 per annum. Provident will also be able to do away with six Primary rate Interfaces (PRIs), which will result in annual savings of $ 500,000.

September 25, 2005

VoIP implementation

eBay’s acquisition of Skype has brought into sharp focus the manner in which companies will choose to communicate in future. Skype adds around 170,000 users daily to its roster of around 55 million users. This implies that people have been quick to appreciate the benefits of Internet Telephony.

Vendors of traditional telephony have also decided to jump onto the IP-telephony bandwagon. In the UK, BT Global Services does business of around £5bn with its corporate clients, out of which £300m comes from calls revenue. The advent of IP may well signal the end of this stream of revenue. Implementing IP telephony will help to cut costs because of the cheaper calls, reduced maintenance bills due to remote maintenance, and reduced mobile bills.

However, it is important for companies to time their migration to an IP system such that they extract the maximum ROI from their existing telephony network. VoIP offers a very distinct advantage if the company has several locations but for a single office, a change to VoIP may need more justification than just reduced charges.

Some companies that have implemented VoIP have also continued with the traditional telephony network, which they hope to phase out once they feel satisfied with the QoS and speeds offered by VoIP

Inmarsat Regional Broadband Global Area Network

Satellite Communication (SatComm) and HowzitOnline.com are two South African companies that have come together to offer a VoIP over satellite product. The device weighs around 1.7 kg and costs around $ 500. The device uses the Inmarsat Regional Broadband Global Area Network (R-BGAN) to make and receive phone calls. R-BGAN offers speeds of up to 144 kbps, which is faster than that offered by terrestrial GPRS. The R-BGAN service covers large parts of Asia, Africa, and Australia. VoIP calls through this service to the US and the UK can be made for less than 5 cents per minute. The BGAN service, which will be launched by the end of the year, is expected to bring down the cost of communication from remote destinations.

Vonage targeted for takeover

Vonage, which was launched by Jeffrey Citron in 2001, may be the next in line for a takeover after eBay’s $ 4 bn acquisition of Skype. Vonage is planning to expand its services in the UK on the basis of an Initial Public Offering (IPO) that it plans to introduce. Vonage hopes to raise as much as $ 600 million through its IPO. However, companies like Google and Microsoft are keen to acquire as much of the rapidly growing Internet Telephony market as they can. This has led to speculations that Vonage may be purchased even before it launches its IPO.

BCM1161

The new integrated VoIP processor chip announced by Broadcom, the Broadcom® BCM1161, is a second generation VoIP processor that consumes low power and provides advanced multimedia functionalities such as a 2 megapixel digital camera, voice and video record/playback, etc. The BCM1161 also allows conferencing and high-fidelity voice capability. It has a single chip that integrates the direct microphone and the speaker interface with the analog voice codec.

SurfUP™ver4

On September 22, 2005, SURF Communication Solutions® announced the launching of SurfUP™ver4 in the market. The product is a universal port solution that supports video transcoding, conferencing, recording, and also has video toolbox capabilities that enable resizing, frame rate change, inserting logos, etc. SurfUP™ver4 offers DSP chip–level solutions as well as DSP–farm resource boards.

The version 4 runs the different media types on one DSP, which is TMS320C64x™ generation by Texas Instruments. The DSP provides scalability to the solution and helps in reducing the Host-DSP bottleneck by supporting UDP/IP, RTP, and H.323. Telecom manufacturers can integrate user-defined channels with the help of the open DSP framework that Surf provides.

As Surf runs voice and video on the same DSP, it provides quality in a cost-effective manner. The product supports dynamic speaker detection, gateway applications, and CTI messaging and recording applications. Moreover, cPCI and ATCA carriers will provide an integrated solution for the Surf PTMC/AMC daughter cards.

September 24, 2005

VoIP security

VoIP undoubtedly has the potential to effect huge savings on call charges, infrastructure, and maintenance. However, it still has to assure IT administrators that it can offer a level of security similar to that offered by traditional telephony. The concerns arise from the fact that VoIP uses Ethernet and is therefore susceptible to DoS, interception, spamming, etc.

As compared to data networks, phone systems may be difficult to patch. VoIP phones are susceptible to Address Resolution Protocol spoofing that can lead to illegal tapping and crashing of the VoIP phone. According to tests conducted by Secure Test, the Cisco 7900 series phones are vulnerable when running the default Skinny protocol and can be crashed easily.

An attacker can use a PC attached to the VoIP network and send a stream of malformed messages that can result in a buffer overflow and crash the instrument. If such an attack were to be performed on a switchboard network, it would be possible for an attacker to disable the network in a matter of minutes. VoIP enables routers are also susceptible to DoS attacks. A large message in which the number of characters exceeds 50,000 can sometimes cause every phone on the VoIP network to reboot.

Prevention of tapping in PSTN networks is more a question of maintaining physical security whereas VoIP data, if unencrypted, can be intercepted by any other phone on the network. It is important that VoIP phones support the secure RTP protocols required to ensure default encryption of data. The attacks on a VoIP network can be carried out remotely with the help of Trojans that may be distributed via a PC connected to the VoIP network.

IP telephony in enterprise voice communications

Enterprise voice communication is shifting from the PBX systems that are based on Time Division Multiplexing to IP telephony. VoIP offers the benefit of convergence to companies as voice, data, fax, video are all managed from a single platform.

Even though converged systems have been available only for some time, they already account for one-third of the market. Avaya and Cisco are two of the leading companies providing IP telephony systems. Although the earlier designs were complex and expensive, advances in the technology have made IP telephony an ideal choice for even the SMBs.

The results of a survey conducted by Sage Research showed that savings and productivity gains are the two main reasons for migrating to IP telephony. With traditional telephony, the initial investment is usually half of the total cost of ownership (TCO). Factors such as relocating or adding instruments over the lifecycle of the communication system lead to an increased TCO. Relocating workspace is far easier with IP telephony deployments. With new IP-based systems, such as those provided by Zultys, phones can be moved without incurring a cost.

Unlike traditional telephony, increasing the capacity of an IP network does not require extensive wiring, a user license may be required but it works out cheaper than installing line cards. TDM systems need additional switches and control units to accommodate growth in user numbers. This installation is expensive and the cost can easily run into more than $10,000. With IP telephony, if the growth rate is greater than the maximum capacity of the base system, the new users can be added by acquiring new phones and the relevant software licenses.

The MX250 system developed by Zultys has features like MXgroup and MXcluster that enable the inclusion of multiple systems located anywhere on the globe. Since traditional systems work with proprietary architecture, adding enhancements such as IVR and call center functionality can be prohibitively expensive. Integration with third party products can be a difficult process. With IP-based systems, most upgrades and enhancements can be achieved via a software download.

The growth of IP telephony will spur competing third party providers to offer services at the most competitive rates; this should make the addition of enhancements even cheaper. IP telephony also means reduced cabling costs as both voice and data traffic travels over the same LAN, which results in the wiring requirements getting reduced by half. The same Ethernet ports serve both computers and phones.

Maintenance costs are reduced with only one set of cables requiring maintenance. Some phones have multiple ports; this allows companies to reduce the number of Ethernet ports. The centralization of the administration process as facilitated by IP-based telephony is allowing companies to shift locations in a cost-effective and timely fashion. Companies that have branches are not required to install switching units at each branch. Each branch has a gateway for providing interoperability throughout the company. The administration is done thorough a user interface that is accessible from anywhere.

IP-telephony maintenance can be done remotely in most cases. Long-distance calls are a significant expense for most corporates. With IP telephony, voice data travels over a WAN on the Internet without incurring an incremental cost. Many businesses use high-speed private WANs. By adding voice and video, they can improve the utilization of the network facilities. As of now, intra-state calls are regulated; even a 20% decrease in IP telephony toll costs would enable a company to recover their investments in toll savings alone.

IP allows for real-time delivery of voice and video data to the various devices. Real-time communication of information enables timely decision-making. IP telephony also supports intelligent call routing and call forwarding.

September 23, 2005

CB1000 voice conferencing platform

The FreeConferenceCall.com and Gizmo Project will be using the CB1000 voice conferencing platform that is provided by Vapps. Vapps is a major vendor of audio conferencing systems used in both IP and TDM networks. The CB1000 voice conferencing platform enables users to hold joint IP and PSTN conferences using either a traditional handset, cell phone or the IP softphone provided by Gizmo Project. The service would be free and available anywhere in the world. voipnews.com reports:

Based on the modular SIPphone VoIP architecture and founded on the principal that next-generation voice communications should be free to all; the Gizmo Project softphone client is a desktop interface designed to place crystal clear, high quality voice calls over the unregulated Internet infrastructure with the ease of instant messaging.

Read More: FreeConferenceCall.com and Gizmo Project Leverage Vapps CB1000 Platform for Free Unrestricted Conferencing Calling

VoIP Inc.

VoIP, Inc. which uses VoIP as its core technology to provide global communication services has launched VoiceOne™ Carrier Direct Program. voipnews.com reports:

VoIP, Inc.'s CEO Steven Ivester commented, "Our Carrier Direct Program enables carriers to increase their level of service offerings and recognize an immediate ROI, utilizing our technical expertise to support them as they quickly gain entry into this fast-paced market."

Read More: VoIP, Inc. Launches its Carrier Direct Program

Infozech

Infozech has been selected by a prominent Mobile Virtual Network Operator (MVNO) in Europe to provide a billing and VoIP solution. Infozech offers a pre-paid VoIP service that includes both IP and PSTN termination. voipnews.com reports:

Infozech’s solution is centered around a billing server and the i-Voice - SIP-based IP calling platform. The PSTN gateway is being provided by Cisco.

Read More: Leading European MVNO selects Infozech solution

Power and cooling solutions for VoIP telephony

In order to supplant the existing telecommunication systems, VoIP telephony has to not only satisfy the QoS mandates but also ensure that the system makes efficient use of power. The patch panels and hubs in the legacy wiring closets will be replaced by routers and UPS that will require sustained cooling to ensure their smooth working. An IP telephony network is made up of layers and has 4 physical locations. IP phones, access layer, distribution layer, core switch, server farm, and call servers make up the different layers. The four physical locations are desktop, wiring closet, main distribution facility, and the data center.

IP phones need around 7 watts of power. The IEEE 802.3af has stipulated that a maximum current of 350mA can be drawn by these endpoints via CAT5 cables. This standard allows for 15 W of power to be delivered to a distance of 100 m. The communication devices are powered either by the data lines or by the wall outlets. In-line power does not require power at the desktop as the instrument draws power from a network switch that is run by a UPS system. If the communication device draws power from a wall outlet, a UPS, with a battery that can run for extended periods, should be provided.

The wiring closets have distribution switches, hubs, patches, routers, etc. Compared to the legacy systems, IP telephony systems use and release more power. Equipment that can be stacked in 1-3 racks can use up to 4000 W of single phase AC power. The power drawn can be at either 120 or 280 VAC and is a function of the network architecture and the switching technology used. Providing circuit breaker protection and the correct receptacles, for example L5-20, L5-30, L6-20, etc is important. A UPS system should be available to protect the system. Factors that affect the choice of UPS include the power requirement, the run time, redundancy to be incorporated, etc.

APC Smart-UPS is a rack-mounted UPS that will ensure 99.99% availability whereas the APC Symmetra RM, which is N+1 redundant, will provide 99.999% availability. Critical applications such as the 911 service may require a higher percentage of availability that may well go into 7 nines. Such requirements can be fulfilled by using dual UPS and dual network switches as well as backup generators.

Rack PDUs should be used only if there is a lot of equipment and the equipment cannot be plugged directly to the UPS. The PDUs should have a meter that displays the power consumption and lessens the possibility of overloading due to oversight. In order to address the problem of cooling the closets, the power dissipation needs to be calculated.

For conditions where the heat load is for less than a 100 W in the closet and the rest of the structure is properly conditioned, wall conduction can provide sufficient cooling. For the same heat load, if there is no HVAC system and the building is not properly conditioned then a small air-conditioner can be installed in the closet. If the heat generated is for greater than 1000 W and a dropped ceiling HVAC system exists with the remainder of the space being conditioned, then to manage the conditioning, the bottom portion of the closet door should be fitted with a vent grill and the rack on which the equipment is placed should carry a hot exhaust air scavenging system.

Important VoIP equipment such as the layer 3 routers and switches is housed in the point of ping (POP), also known as the main equipment rooms (MERs). An MER may have equipment that may use up to 40kW of power and may occupy close to 12 racks. An MER may or may not have a UPS or even sufficient battery backup. In order to ensure 99.999% of availability, a modular UPS system with a backup of at least 30 minutes should be provided. Hot spots can occur at higher density racks; these can be avoided by using air distribution and removal units.

Data centers house sensitive equipment like the application servers and the related software. A large data center may house hundreds of servers that support ERP, WMS, and CRM applications. The data centers can draw more than a 100 kW of 3 phase 480VAC power. The addition of a VoIP network can result in incremental load on the data center necessitating longer runtimes. A separate UPS should be provided for the IP telephony systems that should ideally be housed separately. Redundant air conditioning systems can be installed to ensure higher availability.

VoIP network implementation

VoIP network implementation In order to ensure a high availability of VoIP networks, the following factors must be considered:

Components of call processing: The different components that make up voice systems have different requirements that need to be fulfilled before implementing a VoIP network. The components include voice mail, toll bypass, call center applications, etc. Centralized call processing consists of IP-PBXs, soft phones, etc in centralized locations and the clients can access them through a WAN. Another alternative is on-site call processing, which may be an expensive option if there are too many sites in a WAN. Survivable remote site telephony (SRST) enables centralized call processing but if there is an outage, the phones can still connect via a local router.

Cost: The cost of establishing a network depends upon factors such as the level of redundancy that needs to be incorporated into the network. The network has to be architected keeping the budgetary constraints and ease of use.

Network architecture: WAN and LAN networks are possible. The QoS levels are achieved in different ways for these networks. LANs provide star, bus, and ring topologies. WAN topologies are represented in terms of their technology, examples being Frame Relay, MPLS, etc. LANs use a load-balanced backup circuit in which there are two circuits to share the traffic load. Bandwidth is not a problem with LANs and the load-balancing circuits are relatively inexpensive to install and maintain. WANs use active/passive networks in which the passive network comes into play if the primary circuit experiences an outage. WANs have limited bandwidth to work with and employ expensive circuitry. One has to consider the routes that the circuits will take as these will affect the latency. Network administrators have to ensure that backups provide voice quality similar to that provided by the primary networks. In order to ensure consistent voice quality, the Resource Reservation Protocol should be used for the backup circuit and the utilization of the active circuit should be managed such that in the case of an outage, the increased load on the secondary circuit does not affect the quality of transmission. Another alternative is to route calls over a PSTN network. The choice of backup is a function of costs involved and business considerations.

Cloud Diversity: Carriers such as MPLS, Frame Relay, and ATM are maintained by the carriers. It is safer to have a backup circuit that uses a different cloud than that of the active circuit. This is particularly useful when the entire WAN setup has been provided by a single vendor. Introducing different routings may complicate the network and increase cost but will allow for greater protection against service outages that may happen if a cloud does not work.

Different service layers: Voice and other data traffic should have different access layers. By moving the IP-PBX cluster away from the office network, it is possible to develop an abstracted layer that permits the voice service to continue operating even during maintenance work. The network should be designed after considering the functionalities of the core, access, and distribution layers. Voice virtual LANs provide another layer of abstraction and are often a basic requirement for availing the many proprietary benefits that vendors offer.

DHCP: A Dynamic Host Configuration Protocol (DHCP) provides information regarding the call processing to the VoIP networks. VoIP networks have to strike a balance between the DHCP and DNS service. The DHCP service should be able to cope with an increase in phones in the network.

Subnets: Subnets can sometimes act as points of failure that may affect network performance. Servers and desktops can be given the IP address of the IP gateway either on purpose or by mistake. This can lead to a traffic overflow, which may bring down the subnet. A safeguard is to reduce the size of the subnet, for example one subnet per switch.

Power Consumption: Most networks draw power from the Ethernet (PoE) by using switches. The amount of power drawn should be carefully regulated, especially if large chassis switches are used to provide power to a large number of phones.

September 22, 2005

Cable operators and VoIP

The number of cable VoIP users grew over 900% from 2003 to 2004. The two major companies that between them share more than 75% of the cable users are Cablevision and Time Warner. Comcast hopes to be able to reach out to 15 million homes in America offering VoIP services; it plans to have around 8 million subscribers by the year 2010. Cable companies will have the advantage over VoIP providers as they have control over the quality of the broadband service. In 2004, cable operators spent nearly $ 123 million in purchasing soft switches, routers, media gateways, application servers, etc.

VoIP implementation by cable is going to witness double digit growth till 2007; this is a positive development for companies like Cisco and Lucent. According to the Telecommunications Industry Association (TIA), another reason for the increased spending on VoIP equipment by cable providers is that carriers are moving toward IP networks to manage the VoIP traffic.

VoIP blogs

VoIP blogs are becoming increasingly popular, according to Jeff Pulver who has founded the Voice over Net (VON) conference. In fact, bloggers have their own hierarchy and the good ones are as knowledgeable as industry experts. Companies too have realized the usefulness of blogs in influencing opinions and are evolving definite strategies to make the best use of blogs. voipplanet.com reports:

"The next year will be the year of mega-mergers," said Mark Evans, a senior technology reporter for the National Post who also writes a VoIP blog. "[Subsequent acquisitions] will make us look at eBay and Skype as minimal."

Read More: VoIP Bloggers Gain Influence

September 20, 2005

VoIP QoS II

Along with intserv, another service developed by IETF to fulfill the QoS requirements is diffserv. intserv is not really a built-in service as it requires per hop signaling and reservation of resources along the route that the data is going to take. Also, intserv does not support scalability of a network that is becoming increasingly complex. Differentiated Services or diffserv do not reserve bandwidth but accomplish the transfer of information by using a field in the IP header.

The operations of diffserv are detailed in RFC 2474, RFC 2475, and RFC 3260. Essentially, diffserv is used to distinguish and prioritize between Internet services. This can be done in a variety of ways such as priority levels based on price levels, application requirements, etc; network attributes such as jitter, latency, etc. Network services are provided based on the definitions given by the diffserv architecture. This enables the provision of the available bandwidth to the traffic streams. The packets are classed into specific types for routing purposes.

The packets move from the source to the destination based upon the markings on them. diffserv utilizes the Differentiated Services (DS) field in IPv4 and the Traffic Class field in the IPv6. diffserv uses six of the eight fields that are present in the DS field. The six bits are collectively referred to as the Differentiated Services Code Point (DSCP). DSCP allows for 64 Internet service distinctions. According to IANA, IETF will standardize 32 codepoints and the remaining codepoints will be used equally for local and experimental use and for probable standardized assignments as the need occurs.

VoIP QoS

In the absence of uniform standards, most VoIP vendors have their own solutions that they try to push. Integrated Services is one such solution. Integrated services or intserv was a project that was carried out by the IETF with the objective of "defining a minimal set of global requirements which transition the Internet into a robust integrated-service communications infrastructure." This was intended to enable smooth transfer of audio and video streams in real-time using IP.

IETF concentrated on three issues, namely

• A clear understanding of the services that will be provided.

• A clear understanding of the interfaces for end-to-end routing.

• A clear understanding of any additional requisites for enabling the Internet to support real-time data transfer.

These objectives were explained in three documents, namely RFC 1633, RFC 2212, and RFC 2215. These documents explain the intricacies of the integrated services model and the manner in which the network bandwidth requirements must be met to run the applications.

Resource reservations and admission control are the two major components of this service. They carry out the functions of reserving bandwidths for an application and diverting additional bandwidth to an application, if necessary, but without affecting the other applications. The packet scheduler, the admission control routine, the classifier, and the reservation setup protocol are mechanisms for tracking the flow of data in a network. Packet streams are forwarded by the packet scheduler, which uses procedures such as timers and queues. The packets are grouped by the classifier into a particular class based upon the header details. A decision algorithm run by the admission control allows a router to decide whether the bandwidth can support a new data flow at a given point in time. The details of the flow are maintained by the reservation setup protocol.

VoIP in Japan

According to Kamal Anand, VP Marketing at Meru Networks, Japan represents the most developed and challenging VoIP market in the world. Jupiter Research states that the adoption of VoIP in the US will grow from 1% in 2004 to 12% in 2009. Even though this adoption rate is fantastic, it is less than half of what is projected for Japan. Japan has less than half the population of the US and according to Yano Research Institute, by 2008 will have around 28 million VoIP users.

The main reason for the difference in growth rates is that the level of public awareness in the US regarding VoIP is not very high. According to Will Stofega, VoIP analyst at IDC, the Japanese are accustomed to communication on the move and see VoIP as a cost-effective method of supporting voice services. A call using VoIP is approximately eight times cheaper than that using circuit-switching technology. As far back as 2003, the Japanese government had introduced 050 numbers that would enable the traditional telephone services to make the switch to VoIP.

Another important factor that favors the growth of VoIP in Japan is the availability of broadband. More than a fourth of the Japanese population has access to high-speed broadband and the speeds can touch 12 Mbps, while in the US, the average broadband speed is 3 Mbps. Unlike in the US, regulatory requirements in Japan are not very stringent. Providers are not expected to comply with regulations such as support for 911 and electronic surveillance. Latency and QoS are not major issues in Japan as there is ample bandwidth available for the IP providers.

To try and achieve better growth, VoIP providers in the US should focus more on services and promote features such as soft phone handsets and virtual phone numbers. The entry of online companies like AOL should also boost the growth of VoIP as they have an established customer base to target and can focus more on providing service.

In Japan, VoIP services are marketed bundled along with video and Internet by the cable providers. VoIP is also available with land line and cell phone packages. In the US, providers need to present a stronger case for VoIP apart from highlighting the cost-benefits of VoIP.

September 19, 2005

Avaya makes a foray into the P2P voice market

Avaya has bought Nimcat Networks, which is a Canada-based company in order to enter the P2P VoIP market. It paid $40 million for the purchase. This purchase gives Avaya access to the technology patented by Nimcat, which allows users to avail PBX functionality without actually having to build a PBX. The device knows as nimX enables users to track other users in a network, thereby establishing a virtual exchange.

Avaya plans to add the Nimcat software to its IP telephony solutions in an year’s time. It will develop different products around the technology to suit the needs of different customers. It will also continue to service the existing customers of Nimcat. nimX stands to benefit from the expertise that Avaya has in enterprise-wide security applications.

Spam in VoIP

According to the Yankee Group, VoIP has chalked up an impressive growth record till date. Results of a research by In-Stat show that more than 40% of the larger companies use VoIP. However, this growth is bound to attract the menace of Spam and other security issues. This could actually lead to loss in productivity and expenditure in security tools and their maintenance. Spam in a VoIP scenario is going to occur in the shape of voice messages that will have to be treated in real-time in order to prevent a company’s voice mail system from being flooded with spam messages.

Pierce Reid, V.P Marketing, Oovia, opines that it will take time for VoIP spam to really come to the public’s notice. A more serious threat for the more than 600,000 VoIP phone users is a DoS situation that could occur as a result of too much spam. One way of recognizing VoIP spam is that packets of machine generated messages do not exhibit the randomness associated with human VoIP messages.

Bill for deregulated broadband

A 77-page working draft is being reviewed by the Energy and Commerce Committee in the House of Representatives. The draft is intended to add to the Telecommunications Act of 1996 and incorporate aspects that were not considered in 1996. The bill intends to include voice and video over IP, DSL, wireless, and cable services under the same broad regulatory framework. This will help facilitate the deployment of broadband by states at a municipal level.

The Telecommunications Act of 2005 also was aimed at granting municipalities the right to implement broadband without having to face problems from the providers. This should give a fillip to the $15 million municipal wireless broadband plan in Philadelphia, known as Wireless Philadelphia. HP and Earthlink are the leading candidates for the role of network providers. However, the ambitious plan faces competition from companies like Verizon and Closed Networks that have slashed their rates.

The drivers of VoIP adoption

According to a study conducted by research firm Ovum, the main drivers pushing VoIP adoption are need for communication on the move and cost effective communication. 61% of the respondents in the survey, which was held in England in Apr – May 2005, felt that telecommuting and teleworking were the prime reason for adopting VoIP. 47% of the respondents felt that VoIP enabled mobility of the workforce. 41% felt that VoIP helped in linking remote workers.

According to the survey, companies in England primarily favor a premise based VoIP deployment, with 53% opting for it. Hybrid deployments and hosted deployments had 14% and 12% of the votes, respectively. 32% of the respondents were open to trying out P2P networks as a business-level solution. Even though, the participants in the survey conceded the vulnerability of public VoIP networks, an overwhelming majority said that they were satisfied with their VoIP security set-up. An important fact highlighted by the survey was that several companies evinced a strong interest in wireless VoIP, which is the sign of a maturing market.

A single phone number

VoIP has the potential to leverage the advantages of IP telephony and give people the advantage of a single telephone number. The advantage is of ease of communication, hopefully in a more cost-effective manner. The advent of web-based telephony has made it easier to configure features like simultaneous ringing. VoIP has the advantage of delivering voice regardless of the platform. With VoIP working on IP as the transport medium, it may not even be necessary to have a single number. Different numbers can be configured to respond simultaneously.

The downside to this may be arriving at a tariff that keeps the customer in mind and also does not hurt the service providers. Moreover, not everyone is going to be keen about the idea of getting tracked by a single number. People often prefer to keep their professional and private lives separate. Thus, the idea may receive different reactions from corporates who will probably be enthusiastic about it and from individuals who may view it with some trepidation. This will decide the acceptance of this concept in the mass market.

September 18, 2005

Video on Demand (VoD) has a market

Internet Protocol Television (IPTV) is driving the growth of VoD, which should in five years time become more popular than traditional TV. This is because it will allow people to watch movies according to their schedules.

The development of VoD is going to receive a fillip with telephone carriers like Verizon and SBC taking interest in it. voipplanet.com reports:

Cable companies, according to a report published by the firm Wednesday, are looking at free-on-demand (FoD) as a way to differentiate their service from satellite operators. The research notes that while there has been a 55 percent increase in VoD revenues from 2003 to 2004, revenues continue to lag behind user growth as FoD becomes more popular.

Read More:Video on Demand Taking Its Share

September 17, 2005

TCP/IP and VoIP

The Transmission Control Protocol (TCP) is a connection-oriented protocol. Internet Protocol (IP) and User Datagram Protocol (UDP) are connectionless protocols. In a connection-oriented network, a network has to be established before information can be transferred. In this process, a significant amount of time and resource is spent in signaling. The advantage of this process is that upon the establishment of a network path, attributes of the path such as propagation delay do not change. In fact, connection-oriented networks are also known as reliable networks.

Data networks are examples of connectionless networks in which the data travels in packets. The packets may or may not be delivered to the desired destination. Since the characteristics of the path are not fixed, a connectionless network is often described as unreliable or best-effort. File transfer using FTP, electronic mail using SMTP, and remote host computer access using Telnet were important data applications that were developed using TCP, IP, and UDP. Today, the focus is on applications that integrate voice and data. As these applications are intolerant of delay, they cannot be run on a best-effort network. The answer to the problem lies in adding protocols that will boost the performance of connectionless IP networks. Given below is a list of protocols that enable voice and data transfer over the Internet.

•Multicast Internet Protocol enables the transmission of information from a single source to many recipients.

•RTP Control Protocol checks the performance of the RTP.

•Real-time Streaming Protocol enables data delivery in real-time, which also includes accessing information from media servers.

•Real-time Transport Protocol helps in payload identification and sequence numbering.

•Resource Reservation Protocol ensures that there is sufficient bandwidth available to enable communication between sender and receiver.

•Session Announcement Protocol packets help the end users to make use of open sessions.

•Session Description Protocol allows information exchange regarding the media stream, bandwidth required, session name, etc.

VoIP Service for the Katrina affected

Katrina devastated cities in America got back their voice with PingTone VoIP service provided by Proactive Communications. After the violent hurricane destroyed most telephone and data lines, high quality communication was set up by Proactive with ground based satellite systems using Pingtones VoIP phone service. In spite of lack of power, buildings and transmission lines, deserted family members and aid workers were able to keep in touch with the rest of the world using the service.

Proactive has been providing communication infrastructure to far flung and difficult regions of the world like in Iran and Afghanistan. The portable satellite terminals used are very convenient to carry and can be set up at any place.

The PingTone VoIP service that Proactive uses gives amazing voice clarity and other features. The use of phones by Cisco Systems adds further to its quality. This is the first time that such a service has been used as a part of  disaster relief. The results have been very good and encouraging for everyone concerned.home.businesswire.com reports:

Proactive personnel are able to setup portable satellite terminals just about anywhere local, state or federal first responders need them to: tents, a warehouse or even the back of a truck. These "data networks in a box" give workers and residents immediate phone service, email and Web access."

Read more:Katrina First Responders Deploy Satellite VoIP Services from Proactive Communications

AOL and MSN discuss possible merger

According to the Industry grapevine, AOL and MSN are contemplating a merger in the near future. The proposed deal is believed to be a fall out of the recent purchase of Skype by eBay. AOL, which is a trusted name in the Internet business, has not been able to perform to the satisfaction of Times Warner. Similarly, MSN hasn't added much value to Microsoft. A joint venture between the two should strengthen their existing properties and services.

Both the companies witnessed a hike in their share prices after speculations of their merger hit the market. AOL is also reported to be planning a full fledged VoIP service in the near future. The AOL-MSN union would lead to the combining of their instant messaging services, AIM and MSN Messenger. This would give them an edge in the messaging market, which is warming to some tough competition. Industry Experts believe a synergy between the two will prove to be positive.redherring.com reports:

"I could see the motivation for a deal between AOL and MSN," said Patrick Mahoney, senior analyst with the Yankee Group. "Both are being challenged by Google and Yahoo, and AOL is seeing its dial-up market share going away, so it has every reason to welcome a merger with MSN."

Read more:AOL and MSN May Merge

VoIP standards

VoIP is a new technology and still does not have universal standards. This makes it difficult for network managers to integrate the products obtained from different vendors. In the absence of standards, vendors come out with proprietary standards that make interoperability a difficult proposition for enterprises. This is where the “standards bodies” come into the picture. These bodies are made up of the inventors, developers, vendors who have an interest in a particular technology.

The International Telecommunications Union (ITU) and the Internet Society define VoIP standards. The ITU has its headquarters in Geneva and was established in the 1860’s in order to develop standards for telegraph communications. The ITU is divided into ITU-R, ITU-T, and ITU-D. These are the Radio Communication Sector, the Telecommunications Standardization Sector, and the Telecommunications Development Sector, respectively. The ISDN and ATM standards for telecommunications have been developed by ITU-T. The standards can be identified on the basis of a letter that is assigned to a particular aspect of that technology. ITU-T standards that begin with H relate to audiovisual and multimedia. VoIP is covered under this group of standards. ITU-T standards can be viewed online. The Internet Society has been involved in issues related to the Internet since 1992. It concentrates more on packet switching and data transmission issues.

The Internet Society also works as small groups such as the Internet Architecture Board (IAB), Internet Research Task Force (IRTF), Internet Engineering Task Force (IETF), etc. Internet Standards, also known as Request for Comments or RFC documents are developed by the IETF. Some well-known RFCs include the Hypertext Transmission Protocol (HTTP), RFC 2616, and the Session Initiation Protocol (SIP), RFC 3261. Organizations such as the American National Standards Institute and the European National Standards Institute also influence standards but at a less broad level.

A company that wishes to implement VoIP should try and get an understanding of the standards that govern their VoIP devices and applications. This is because applications that follow the ITU-T specifications may have different networking and architecture issues than those that follow the IETF standards. Knowledge of the standards will help in making the right product decisions and also help to solve interoperability issues.

Instant Messaging and VoIP

By the end of the year 2005, IM providers will have around 850 million accounts and a lot of these will also have access to VoIP. Keeping this in mind, companies such as Yahoo and AOL are offering VoIP-over–IM tools to enhance user experience. According to experts, this may have major implications for companies like Skype and Vonage. Even though, voice facility in IMs has been around for some time, the connectivity has been mainly PC to PC and not of a very high quality.

A new version of AIM, known as Triton, and a new Yahoo client have been released. Both have laid an emphasis on voice and have reported an increase in voice usage since the introduction of their beta products. The improved quality of codecs has helped to better the quality of VoIP over IM. Skype, which uses the GIPS codecs, took the lead in popularizing high quality VoIP communication. Triton uses a range of codecs including GIPS. Vonage and Yahoo use the Xten codec for their softphones. According to Matthew Anderson, analyst at the Radicati Group, even though Skype is the most popular VoIP software, it has not had a serious effect on IM usage. voipplanet.com reports:

The new generation of IM clients, offering enhanced voice capabilities, amongst a myriad of other advanced features, means that users don't need to leave their IM client.

Read More: VoIP over Instant Messaging? It's Coming—and It's BIG!

VoIP subscribers for sale

Total Marketing One (TMOne), which is an Iowa-based marketing company, is providing VoIP subscribers to providers. VoIP providers can get subscribers from TMOne in batches of 20,000 at a rate that can even be $200 per user. The rate is negotiable and depends upon the number of users that the provider wants and the brand equity of the provider. TMOne ensures that the subscribers stay with the providers for at least three months or for the money-back period, whichever is more.

According to Anthony Marlowe, CEO, TMOne, with an agreement reached on the 911 issue, VoIP is poised to make an impact on the mass market and by 2009 there could be as many as 40 million VoIP users in the US. This figure is bound to attract big players and MSOs into the VoIP market. The cost of buying users from TMOne is less than half of what companies would pay if they were to hunt for subscribers on their own. Moreover, with VoIP growing, these subscribers can be sold further down the line at a price which could even be 10 times their purchase price.

TMOne attracts customers through different mediums, chiefly via telesales, print, retail agents, etc. However, around half the customers are obtained through telesales. TMOne maintains a comprehensive database of around 30 million broadband users. The database includes 20 million home users, 2 million SOHOs, and around 8 million SMBs. Companies like TMOne have to be careful that they do not try and hardsell a provider to a prospect so much that the prospect develops a prejudice against that provider. Also, given the fact that there are too many providers offering VoIP services, it is imperative that TMOne closes the sale in favor of its client as soon as possible.

September 16, 2005

VoIP affects network performance

According to a survey by Enterprise Management Associates that covered 100 companies, it was found that VoIP performance was affected by the fact that it competed with data applications for bandwidth. Close to 90% of the respondents who were polled in the survey felt that having VoIP performance monitoring capabilities was critical in ensuring smooth functioning of VoIP. The survey reinforced the thought that although VoIP is being rapidly adopted by industries, its scalability can be hindered by the absence of management policies to regulate the effect of VoIP and data communications on network performance.

According to Jim Vale, Product Manager, Network General, most companies know that they have to fulfill rigorous performance criteria to ensure real-time VoIP transmissions. Yet, they do not seem to grasp the significance of the effects that VoIP implementations can have on the network. This can handicap them in their efforts to manage "mission critical" applications that run on the network simultaneously with VoIP. The fact that VoIP is not like other applications that run on TCP/IP and its transmission requirements are different makes VoIP deployment a slightly tricky issue. VoIP transmissions are high priority and have poor tolerance for dropped packets and retransmissions.

Network General and Fluke Networks are two companies that are working toward developing network performance tools that can be integrated with the regular network management activities as a part of the system. Network General has focused on the L2-L7 protocol analysis and provides tools for application-level analysis. The VoIP Lifecycle Solution offered by Fluke Networks manages and troubleshoots voice as well as data traffic.

VoIP in educational institutions

The Robertson Education Empowerment Foundation (REEF) enables universities around the world to communicate via VoIP as a part of its Global University Phone System (GUPS) project. The universities will receive the necessary hardware and software that includes the Asterisk IP-PBX, which runs on Linux. This will enable communication within the university and between universities. SIPphone will extend technical support free and will also provide a directory service that will help in routing the calls correctly from PC's to phones.

UC San Diego, University of Philipines, UC Irvine are some of the universities that are a part of the GUPS scheme. GUPS plans to connect all the 12,000 major universities worldwide. GUPS has managed to connect universities that are using PBXs manufactured by different vendors such as Cisco, Ericsson, etc. Asterisk was preferred over SIP-specific IP PBXs as it functioned well with the T-1 adapter. The use of SIP and Asterisk by GUPS should educate people on the working of open standards, which should help in the spread of VoIP.

September 15, 2005

VoIP semiconductors record impressive sales

The increase in VoIP adoption has led to a growth in the consumption of VoIP semiconductors, which has encouraged manufacturers like Texas Instruments to concentrate on releasing new products and companies like Broadcom Corp. have recorded impressive growth rates. The new generation of Ethernet IP phone designs will have inexpensive, single-chip solutions with hardware-based security.

According to a survey by IDC, the market for VoIP semiconductors recorded a growth of 40% in the year 2004. Texas Instruments recorded impressive sales of its Digital Signal Processor (DSP) solutions. Mindway made its presence felt on the basis of its growth in media gateway silicon. Six top IP phone manufacturers purchased IP phone solutions from Broadcom.  Broadcom received a fillip in sales with its Gigabit VoIP chip.

Texas Instruments is in the process of creating a voice-over-cable chipset and a DOCSIS reference design. The new chipsets being designed have low-bit-rate codecs that promote efficient use of bandwidth and faster broadband speeds. The Wideband DOCSIS 3.0 reference design will help to improve the performance of VoIP and video streams as it incorporates several up and down streams in a single codec. Marvell will be introducing system-on-a-chip VoIP solutions that will cater to residential VoIP gateways as well as the ultra-low-power VoWAN handsets. The low-power circuit systems by Marvell have features such as WLAN encryption, QoS technology, and WPA/WPA2 security.

VoIP equipment sales continue to grow

An important indicator of the impact of VoIP is the increase in the sales of VoIP-related equipment such as IP PBX. The sale of Carrier VoIP equipment has increased more than 50% in the period Apr 2005 - May 2005 as compared to the year 2004. The sale of $614 million in the second quarter was an 18% increase over the first quarter.

According to Infonetics, VoIP should achieve a 40% penetration of the market by the year 2008 and the sales revenue should be approximately $6 billion. The growth would need an increase in the number of VoIP subscribers from the current numbers to approximately 25 million subscribers in the US. The growth of the IP PBX market coincides with the decline in the sales of TDM equipment. The worldwide sales of PBX/KTS units increased to $1.6 billion for the period Apr 2005 - May 2005 and by the end of the year the global PBX/KTS sales should touch $7 billion.

For the second quarter of 2005, even though TDM units at 49% represent the maximum numbers in terms of sale, they account for only 26% of the revenue. Hybrid units and pure IP accounted for 42% and 9% of the total units sold, respectively. They accounted for 58% and 16% of the sales revenue, respectively. 

The revenues from TDM sales are expected to wane to $759 million by the year 2008. By then, the sales of pure IP PBX and hybrid systems should grow four times and three times, respectively.

VoIP implementation and security concerns

The implementing of VoIP applications and the associated security concerns are leading to an increase in the sale of security applications, according to a research by In-Stat. According to the research, companies become more aware of the security aspect after they have installed VoIP. More than 60% of the respondents who stated their security apprehensions had already installed VoIP. Around 12% of the respondents who had security apprehensions were still in the planning stage.

Another fact that the research threw up was that companies with an employee strength of 500-1000 employees were more concerned about the security and many planned a security overhaul next year. This is because VoIP is susceptible to all the threats that currently plague the data networks, these include worms, viruses, spam, etc. Apart from these threats VoIP communication can be intercepted, the application be used for perpetrating a fraud, and sensitive information accessed via the call logs.

VoIP security is inconvenienced by the fact that there are no fixed standards and therefore the vendors sometimes find it a little difficult to add call-control security protocols.  Traditional security application vendors are updating their existing products to provide VoIP security. SonicWall and Checkpoint have upgraded their firewall applications to support VoIP. TippingPoint, Borderware, and Ingate offer point products aimed at satisfying VoIP security protocols. Companies such as Cisco and Juniper have acquired other companies in order to support their converged networks.

The growth of VoIP will help the security appliance market to cross $7 billion by the year 2009. The availability of voice and data on a single network will result in a change in the manner in which network security is approached, there will be a shift from defense equipment on to the perimeter toward intelligent management of the traffic.

VoiP considerations for homeowners

One of the main attractions of VoIP is its low cost. Before homeowners decide upon switching to VoIP, there are a few things that they should consider.

• The compatibility of their high-speed connection to manage IP calls should be tested. TestYourVoIP.com provides a free test to this end. Also, it is safer to switch to VoIP on a secondary line and consider switching the main line if the results are satisfactory.

• VoIP service providers offer different packages. Consider your requirements before opting for one. Elementary schemes with bundled long-distance minutes can be availed for as less as $10. Business packages may cost more but they offer services such as conferencing, fax, etc. A soft phone service is a good option if you travel with your laptop. Also, many providers offer unlimited long-distance calls, but only to select countries. Calls to Canada are treated by most as domestic long-distance.

• Services like AT&T offers AT&T CallVantage that searches for the call receiver by ringing up to five different phones as a part of its call forwarding service. A large VoIP company may charge $10 more for its service but it offers the guarantee of continued service, which may not be the case with tiny startups liable to fold without notice.

Security threat to Wi-Fi by "Evil Twin"

Evil Twin is new threat to Wi-FI users. It refers to the use of malicious servers that pose as genuine ones and try to extract sensitive information such as credit card numbers and bank details. The attack can be carried out by a person close to a hot spot.

The malicious server interferes with the signals sent to the wireless users. The users are tricked into logging in to the fake server. Evil Twin has special significance for countries like the UK and US as they have a very high concentration of Wi-Fi hotspots. The UK has more than 9,000 hotspots whereas the US has more than 22,000.

The growth of Wi-Fi has been helped by the Centrino chip, which now comes with additional security features and a built-in support for Cisco-compatible extensions. T-Mobile has a network of more than 4,500 hotspots across the US and it is implementing authentication based on 802.1x in order to prevent security breaches.

The implementing of a strong security network assumes special significance because of the variety in which the breaches can occur. Attackers can launch man-in-the-middle attacks and can capture data without even requiring a cellular card. These attacks are more likely to occur at public hotspots as corporate VPNs are generally more secure; however, a company employee can expose himself to risk if he tries to access a corporate network via a public hotspot.

Corporates are faced with the problem of rogue access points that can spring up anywhere in the company premises. The problem lies in the fact that any network to which a Windows user has connected to in the past gets reconnected by default. A patient attacker has only to wait long enough for a user who has previously networked with him. As with any technology that gains currency, the first step in the defense process is to educate the user.

Aspects of VoIP security

According to Varun Nagaraj, V.P, Product Development, Extreme Networks Inc., from a security point of view, it is a better idea to opt for a two-tier network than the current three-tiered networks. A two tier network architecture provides continuous uptime and more robust security. The two tiers consist of a core network and a unified access tier, which faces the user.

Extreme has launched the Aspen 8800 Series of enterprise LAN switches that enable a sturdy edge network that provides greater performance than the currently available edge switches. The switches also assure greater availability as they provide management modules with automatic failover and redundant controller boards. The Aspen 8800 Series comes with a ten slot and a six slot chassis. A module with 48 ports of 10/ 100/1,000BaseT, POE (power over Ethernet) is also provided in order to extend better support to the wireless access points. 

September 14, 2005

Predeployment testing for better VoIP implementation

Two factors that can affect the time required for deploying VoIP and the extent of its usefulness include preassessment and periodic in-house assessment and monitoring. Preassessments are carried out with the objective of ironing out the faults, such as poor cabling, bandwidth congestion, etc, that may not affect data transfer but can be detrimental to real-time communication.

Usually, corporates employ consultants for carrying out objective preassessments; however, in-house administrators must team up with the third-party integrators to develop their own skill sets.  ClearSight Analyzer is a voice monitoring tool that tests the quality of the actual voice traffic. AppareNet Voice is a tool that provides end-to-end assessment by generating simulated traffic. This allows the system administrators to find out the trouble zones in the network before starting to implement VoIP. The network is assessed for jitter, delay, and packet loss and graded by using a Mean Opinion Score (MOS) or an R-Value score.

AppareNet Voice uses Internet Control Message Protocol (ICMP) that queries the IP addresses on the network and by routing some sample traffic to the remote hosts, it is able to give an assessment of the network performance. Vivinet Assessor simulates the call load over the distributed endpoints and provides a performance report.

The MOS value ranges from 1 to 5, with 1 being not fit for use and 5 being excellent. It is a subjective score and is based on user perception.  The R-Value is an objective score and is based on the G.107 specification. It has a scale of 1 to 100 that measures loudness, signal clarity, and signal disturbances.

FCC regulation for tapping VoIP calls

On August 8, 2005, a new rule was announced by the Federal Communications Commission (FCC) with the aim of broadening the scope of the Communications Assistance to Law Enforcement Act (CALEA). The law would require Internet broadband and VoIP providers to facilitate wiretapping for law enforcement agencies.

The ruling has come about because of the increase in the amount of online communication between terrorists and the need for security on all fronts post 9/11. The rule is also a result of the growth of VoIP as a means of telephony and the fact that it is increasingly replacing circuit-switched networks. eweek.com reports:

Experts said it is hard to argue that a VOIP message from one terrorist to another exhorting the destruction of the Met Life Building, or another landmark, in New York City, is First-Amendment protected free speech.

Read More: Is VOIP Wiretapping a Privacy Threat?

One million customers for Vonage

Vonage has crossed one million customers, it made this announcement on the 6th of September, 2005. The company is based in Edison, N.J and its achievement is an indicator of the rate at which the VoIP industry has grown. Skype Technologies, based in Luxembourg also has over a million subscribers but a majority of them subscribe only to SkypeOut, which is a service that enables calls to be made by using Skype. In order to receive calls, customers have to purchase SkypeIn. pcmag reports:

Both Cox and Comcast said they each have more than 1 million phone subscribers, but the vast majority still use a network of circuit switches rather than VOIP technology. Time Warner Cable, which uses VOIP exclusively, has about 614,000 subscribers. Cablevision has about 250,000 VOIP subscribers.

Read More: Vonage Reaches a Million Subscribers

The growth of residential VoIP

According to researchers Frost and Sullivan, residential VoIP is ready for the mass market. The difference between the early adopters and mass-market customers is that mass-market customers are not usually driven by the lure of a new technology. For them, it is the service that counts. With respect to VoIP, it means that the customer must be assured of the QoS, secure communication, and access to 911. If these requirements are met, VoIP can increase its presence as it offers its service at a low rate and the availability of broadband facilitates communication between different devices.

Also, along with its low cost, VoIP offers features such as virtual numbers, video conferencing, text messaging, click-to-dial, etc. These options are sources of potential revenue for the customer. Features such as Unified Messaging (UM) allow users to manage voice, text, and fax messages in one box and share them with other devices. The number of VoIP lines is expected to grow to 18 million by 2010 and the market is expected to grow from $295 million at present to $4 billion by 2010.

The implications of eBay acquiring Skype

eBay's multi-billion dollar purchase of Skype has puzzled some industry watchers who are wondering aloud about the usefulness of the deal to eBay. However, according to some, it makes perfect business sense for eBay that gets a database of around 54 million subscribers of Skype and also the $60 million revenue that Skype has projected for the year 2005. eBay purchased Skype for a sum of $2.6 billion. eBay can now leverage its newly acquired VoIP capabilities to provide a whole new range of services to the consumer. By acquiring VoIP capabilities, eBay has joined the ranks of Google Inc. and Yahoo Inc.

According to Maribel Lopez, who is an analyst with Forrester Research, voice is no longer an isolated application, it figures prominently in everything that an individual does online and will play an increasingly large role in the services that online businesses will offer in future. According to some industry watchers, by acquiring Skype, eBay has given itself the option of opting for a portal-like business model ans increase the range of services, such as better e-commerce and customer service tools.

By acquiring VoIP capabilities, companies such as eBay can aspire to become the center of their customers' web browsing experience and help in online trading by providing video capabilities. According to Hani Durzy, spokesman for eBay, the company looks at voice capability as being ubiquitous in its scheme for offering better service in the areas of auctioning cars and industrial tools. Also VoIP will help to better the communication that occurs during the buying and selling of antique items and other unique pieces.

However, according to http://www.eweek.com/article2/0,1895,1858398,00.asp, What is eBay thinking?, an article by David Coursey, acquiring Skype may not be such a good purchase for eBay, especially at the price that it paid for Skype.

September 11, 2005

What is Wavigo

Wavigo is a peer to peer protocol that supports instant messaging and VoIP and offers a host of features like SMS messaging, media player, access to RSS feed, etc. Wavigo allows its users to connect to Skype as well as to chat with Yahoo, MSN, and ICQ users at the same time. Wavigo's features can be personalized to access news and stock quotes and the user can send SMS messages to multiple recipients. Wavigo supports Podcasts and users can manage their own audio and video playlists as well as access streamed audio/video on Wavigo. Wavigo connects with Skype only if Skype is also running on the PC. The major advantage of Wavigo is that acts as a single interface for executing IMs to almost all such services. However, the Wavigo interface is not as appealing in its looks and reaching out to the tabs can be a little tedious. Wavigo runs on broadband as well as dialup and in the coming months it will get connected with regular telephone networks as well.

What is a VoIP Peering Fabric

Voice Peering Fabric (VoIP), as launched by Stealth Communications, is a private voice Internet that enables the users to communicate via peer-to-peer connections and also functions as a junction for exchange of VoIP traffic. If VPF gains popularity, its users can hope to route their traffic without having to depend on the PSTN providers. The VPF application service providers allow the customers to execute transactions in a transparent manner. VPF includes features like caller ID, LNP, 800 services, etc.

A single VPF ethernet connection is sufficient for managing access to multiple services and allows companies to do away with separate TDM connections. VPF has an ENUM database registry that limits the VoIP calls to within the IP domain on the VPF. The VPF ENUM database conforms to the IETF ENUM standard 3761. This allows VPF interoperability with devices made by companies such as Cisco and Nextone. However, according to Joe Laszlo, Analyst at Jupiter Research, VoIP peering as offered by companies like Stealth Communications and Arbinet will not pose a serious threat to traditional telephony for quite some time.

Causes of jitter and methods of jitter measurement

Jitter is the variation in the transit delay that packets experience while traversing a network. It is caused by queuing and the serialization effects on the packet path. Class based queuing, reserving bandwidths, high speed links like SDH and E3/T3 are some of the QoS initiatives that will help in controlling jitter.

Jitter is of the following types:

Constant jitter: In this, the variation in delay is more or less constant.

Transient jitter: An unnatural incremental delay, sometimes only by single packets.

Short term delay variation: It occurs due to changing routes and exhibits increasing delay for some packets as well as an increase in packet to packet delay.

Examples of delay

System packet scheduling delay: It is a transient jitter. VoIP with soft phones often experiences jitter as more than one program may be running on the CPU, thereby slowing it and transmission time jitter is introduced.

Congestion in the Local Area Network: It is a transient jitter and occurs for short durations. It is governed by the maximum back-off time and the delay between packets. If the LAN cannot be contacted by the VoIP endpoint and the back-off time limit is reached or if another packet is ready for transmission, then the previous packet may be dropped. A 10 Mbit ethernet has a high back-off time as compared to the VoIP packet spacing and hence the jitter limits are governed more by the packet spacing and are usually in the range of 10-30 milliseconds.

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