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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.

Firewall routers: It can lead to a transient delay as well as short term variations. Firewall routers such as double socket routers reestablish an IP flow on the inner side of the firewalls after they have terminated it on the outer side. This helps in regulating the payload that gets forwarded to the inner networks. However, this leads to variable delay.

Access Links: These lead to short term variations and are often responsible for jitter as they constitute a bottleneck in the network. As ISDN and cable modems have bandwidth problems, the jitter introduced due to access links can be severe, sometimes up to 30 milliseconds of delay for each packet.

Load Sharing: Load sharing between IP service providers can lead to a constant jitter. Sometimes, multiple access links are routed through one IP service provider and this can lead to jitter if the delays across the links differ.

Load Sharing by an IP service: It can lead to a constant jitter. When IP service providers route traffic over more than one internal route in order to even out the load on the network, the difference in delay on each route can lead to delay.

Load Sharing within routers: It results in a constant jitter. When routers process packet in multiple queues in order to boost router capacity, it can lead to low levels of jitter. In order to support high capacity some routers employ a multi-processing approach in which packets are processed by multiple parallel queues. This can introduce low levels of jitter due to short term differences in queue size.

Routing table updates: These can lead to transient jitters. Routers perform periodic updates in order to ascertain packet priority and dispatch the high-priority packet first. This can lead to a delay in the transmission of some packets and sometimes some packets can experience very high delays.

Route Flapping: This causes transient jitters and can be traced to varying levels of congestion and link breakdowns. Route flapping occurs when a routing table is updated and is characterized by a low frequency oscillation.

Timing Drifts: It causes transient jitters and can result in "jitter buffer events", in which the buffer can either be overfilled or it has excess capacity. The timing can be reset if an NTP server is used.

A few approaches used for measuring jitter have been described below:

RTCP jitter is measured in terms of packet to packet delay. If we consider the delay between two consecutive packets to be Ta and Tb, then the variation is represented as abs(Tb-Ta). The mean of the packet to packet delay variation can be given by MPPDV = mean( abs(ti – ti-1) ). The MPPDV in this case represents the jitter levels in scenarios in which the packets arrive early and late in an alternate fashion.

The mean absolute delay variation (MAPDV) for a packet having a nominal arrival time of ai is given by mean(abs(ti - ai) ). In case of a route change, the value may not be an accurate estimate of the jitter buffer size or discard rate. Jitter buffer behavior can be understood in a better manner by considering the MAPDV in relation to the average or the adjusted value.

An alternative approach is to determine the mean absolute packet delay variation with regard to a short term average or minimum value – termed here the adjusted absolute packet delay variation. This can provide a more meaningful relationship to jitter buffer behavior.

If the nominal arrival time (denoted below ai) for a packet is known or can be determined then the absolute delay variation is abs(ti – ai).

The mean absolute packet delay variation is therefore:

MAPDV = mean( abs(ti – ai) )

This value can be misleading if a delay change occurs (e.g. route change), as a constant offset would be included. As even fixed jitter buffers can adapt to delay shifts, this means that the reported jitter value would not necessarily be a good indicator of ideal jitter buffer size or discard rate.

In a given time interval, the difference between the minimum and maximum transaction delays is given by IP Delay Variation. The time gap between successive measurements has an effect on the IP Delay Variation readings.

A Time Series Analysis is an alternative method in which sequences are fed into a filter function and matched with the data that is being modeled, for example with a Moving Average filter function. The jitter can be modeled as the sum of the processes that occur randomly. This will help in understanding the time varying nature of jitter and can also be used to emulate the jitter in IP networks. By modeling the jitter buffer operation, it is possible to estimate the packet losses in a live scenario. This helps to speed up the measurement process as there is no need to relate the jitter metric to a discard rate.

Jitter buffers are used to reduce jitter from the voice stream; however, in the process of reducing jitter, the buffers can increase delay and packet loss. Jitter buffers are either adaptive or fixed. Adaptive jitter buffers can vary their size as per the amount of traffic. Jitter buffers can adjust automatically with the delay in traffic, this permits the data to be retained for maximum time before it has to be discarded. The jitter buffer is sensitive to the recent minimum delays and is aware of the maximum permissible delay. This helps it to adjust to any changes in delay.

An increase in jitter levels or the presence of a discard event is a trigger for adaptive jitter buffers to react. In the presence of a discard event, the jitter buffer size is increased. For jitter events that happen close to one another, an adaptive jitter is preferred; however, for jitter that occurs over a period of time, as in a LAN, increasing the size of the jitter buffer may lead to delay. Jitter modeling should be such that IP network emulation can be carried out with the help of data obtained by using a time series model. The impact of jitter can be measured on a VoIP service by using a jitter buffer emulator that can deduce the number of packets that will be discarded.

September 10, 2005

Packet loss due to burstiness

Packet loss in IP networks happens in bursts and there are various models that analyze the cause for packet loss and bit errors; usually it is congestion in the network and jitter.

Bernoulli Model: It is a popular independent loss channel and assumes that packet losses occur with a probability Pe. Therefore, if the number of packets in the network is N, the number of packets lost is N.Pe.

Gilbert Model: It is a well-known and widely used burst model. The model has two states, these are a gap state with a value 0 and a burst state with a value 1. The gap state is a zero loss state and the burst state is referred to as a lossy state.

Markov Model: It is a multi-state model and the system keeps oscillating between states. Short term dependencies between lost packets can be studied with help from the 2-state Markov model. It can be combined with the Gilbert-Elliott model to understand consecutive losses as well as the longer events that occur due to link failure and can last for more than 10 seconds.

VoIP quality suffers due to bursty packet loss even if the loss rate is as low as one percent. This is because the packet loss occurs in short and dense bursts that leads to degradation of sound.

VoIP Telephony, The Need of the Hour

In a world which is prone to changes in technology, VoIP is seen as a major invention. The unprecedented growth of VoIP users has proved that VoIP is going to dominate the telephony market in the coming years. With the announcement of Vonage, a leading VoIP provider that it has reached one million customers is pointed at the trend for the future. Vonage is not the only company, which is expanding its VoIP network. Many other companies believe that VoIP is the need of the hour. They are confident that providing VoIP services to the customers, they can satisfy all their requirements.

It is beyond doubt that consumer VoIP is on the rise. The mainstream telephone users are now eager to avail low cost Internet telephony and the improved features which are provided by VoIP. news.ft.com reports:

Unlike most traditional phone calls, calls based on VoIP technology are digitised, chopped up into tiny electronic packets and then sent to their destination over the public internet. That translates into more efficient use of bandwidth and lower costs for VoIP service providers.

Read More: Why VoIP telephony is quickly coming of age

China Telecom may block VoIP

China Telecom is exploring ways to block VoIP phone calls offered by major VoIP providers like Skype. China Telecom is China's largest fixed-line operator. Skype's free software allows people talk for free over the Internet using computers and microphones. It also can be used to call landlines for free.

Such free services threatens the business of fixed-line phone operators. China Telecom wants to prevent users in China from logging on to Skype's server. It is also trying to monitor and control online data volume. If someone makes a phone call over a China Telecom broadband connection, it will be disconnected. news.moneycentral.msn.com reports:

An operator at Shenzhen Telecom -- a branch of China Telecom in the southern city of Shenzhen -- said Saturday that downloading software for voice over Internet calls is not allowed by Shenzhen Telecom. She refused to give her name. Operators at Beijing Telecom and Shanghai Telecom -- other China Telecom branches -- said they had heard of no such restrictions.

Read More: China Telecom Seeks to Block VoIP

Bell is all set to make a mark in VoIP market

To counter the competition by the cable operators, Bell has launched digital telephone services. Bell is Canada's largest phone company. It launched Bell Digital Voice in Hamilton and Canada. Bell Digital Voice is a substitute for traditional phone service. It has several new features which will be helpful for the consumers.

This move is seen as very significant keeping in view the growth of VoIP in the telecommunications world. Cable companies such as Vonage has wrested business from Bell over the past few months by offering VoIP services. Now Bell is in a position to retain its customers and at the same time it will attract more business. theglobeandmail.com reports:

The Bell offerings include many bells and whistles such as voice-mail to e-mail as well as competitive pricing. Bell dipped its toes in the VoIP consumer market in March by launching the so-called lite version of the service in three towns in Quebec.

Read More: Bell plays catch-up on VoIP

Mobif plans to sell 50,000 new VTalk Analogue Telephone Adaptors in Asia

VoIP product maker Mobif Bhd is planning to sell 50,000 VoIP products all over Asia by the end of the year. Mobif had launched VTalk analogue telephone adaptor in April. so far, it has sold 25,000 untis of this phone.

The analogue telephone adaptor is based on VoIP technology that uses broadband Internet connection. This converts voice signal from the telephone or computer into a digital signal that travels over the Internet. It then converts it back at the other end so one can speak to another with a regular telephone number or computer. The cost for such calls is either free or is at the reduced rates. Besides calls, the adaptor also offers additional features such as three-way conferencing, sending of voice mail to email, online itemised billing and three-way video calling. biz.thestar.com reports:

The adaptor is similar to a two-way PABX (private access branch exchange) system but enables borderless connectivity and mobility. It is suitable for home office users and travellers who need to make long distance calls.

Read More: Mobif targets sale of 50,000 VoIP products

Carrying voice over frame relay, IP, and ATM - Part 2

Voice over Internet Protocol (VoIP) uses IP, which is a connectionless protocol. IP facilitates efficient bandwidth allocation as packets do not follow a preallocated path between endpoints. Paths that are open and are less congested can be used for transmitting the packets. To ensure a high QoS level, it is preferable that the packets get transferred on the same path. Headers used in IP traffic consume bandwidth because of their size, they can be up to 20 bytes, headers in Frame Relays and ATM cells are of 2 bytes and 5 bytes, respectively. The headers contain information that ensures the arrival of the packets at their desired destination as well as in rearranging the packets at the receiver's end.

Fragmentation, jitter buffering, prioritization, voice compression, silence suppression, and echo canceling are some of the methods used in an IP network to increase bandwidth efficiency.

Prioritization: Prioritization is closely linked with QoS. At present, there is no widely accepted QoS standard for IP services. RSVP was an IP QoS protocol under which a sender could try and obtain permission to dispatch his data in a particular manner. It has led to the development of the Differentiated Services Model that uses Type of Service (ToS) to determine the type of traffic at the gateway between the user and the service provider.

Fragmentation: It is carried out to minimize the delay of voice traffic and is performed in a similar manner as in Frame Relay. However, this leads to an increase in the number of IP headers, which means that IP voice traffic may require up to 50% more bandwidth than Frame Relay voice traffic. Improvements in header compression and router technology should help in minimizing bandwidth consumption in IP.

Voice compression: As voice traffic usually travels over links that do not have a very high speed, for example VPNs at many SMBs run only at 28.8 kbps. The ITU G.723.1 standard supports voice compression over IP for dial-up modems and ensures toll quality voice.

Jitter Buffers: These store the packets that arrive so that the delay in the variations is minimized. The setting of the buffer can affect the quality of the conversation. The maximum size of an adaptive jitter buffer can go up to 100 ms to 200 ms. The ideal size is between 30 ms to 50 ms.

Echo Cancellation: Echoes occur because of a mismatch in the impedance in the circuit-switched network or a faulty coupling between the microphone and the speaker of a telephone. VoIP networks can face greater delays than circuit-switched networks and consequently require better echo cancellation techniques. G.165 and G.168 are some of the specs recommended to counter echoes.

Silence Suppression: It is also known as Voice Activity Detection (VAD) and implies the ability to refrain from sending audio packets on an RTP stream during the silent periods, which include the pauses between words and the natural pauses in a conversation. Silence Suppression can help in reducing the bandwidth requirement by 10%.

Voice over ATM: Asynchronous Transfer Mode is an ITU-T standard that lays down the specifications for cell relay of information such as voice, video, etc. The information is relayed in small cells of a fixed size. The technology has the advantage of being high speed and scalable. However, it is an expensive technology. ATM is being increasingly used by corporates to transfer large amounts of voice, graphics, and other such information. ATMs use fixed-size cells that consist of 53 octets/bytes each. The cell consists of a header and a body, with the header consuming 5 bytes and the body taking up the remaining 48 bytes. The small packet size makes ATM suitable for transferring voice and video data as these data types require a steady flow and large data packets take time to download.

Fragmentation: The ATM network uses high-speed switches to run the data through its course. This is possible primarily due to the fragmentation that is built-in into the network. ATM networks use high bandwidths that help in minimizing congestion problems and ensure reliable delivery of data packets. This helps the ATM providers in delivering a high QoS.

Prioritization: VoATM follows the standards laid down in the Adaptation Layer 1 (AAL1) protocol as per the Constant Bit Rate (CBR) service. CBR provides Circuit Emulation Services (CES) that transmits a continuous stream of information; this enables the network to apportion the desired bandwidth to a connection for the transmission period. However, as with circuit switching technology, the voice quality that comes from a regular transmission comes at the price of efficient bandwidth utilization. On occasions, CES can transmit semi-filled cells instead of waiting for the cell to fill. This can lead to wastage of up to 20 bytes of bandwidth per ATM cell. Dynamic Bandwidth Circuit Emulation Service (DBCES) is similar to CES except that it transmits only when the receiver is off the hook.

A Variable Bit Rate (VBR-RT) service as specified by AAL2 is the accepted standard for VoATM. Packets of size 1 to 64 bytes can be transmitted by following the AAL2 standard. These packets are also known as minicells and can be incorporated into an ATM cell. AAL2 supports a variable payload, which helps to improve bandwidth efficiency. AAL2 also supports voice compression and silence suppression and enables multiple voice channels over one ATM connection.

Interoperability between networks: Achieving interoperability between the various networks will allow users to benefit from the best that each network has to offer. ATM offers a high QoS and a good speed, Frame Relay provides an installed base, and IP has a global reach. The extent of compatibility is limited by the prioritization methods and signaling protocols, even though these networks follow similar fragmenting techniques. The level of interoperability will increase with the introduction of standardizations within the protocols, which will facilitate the interworking.

Currently, the Frame Relay Forum has set standards for transmitting voice over Frame Relay; however, there are no standards for voice switching between VFRADs. This has led to the development of proprietary solutions that limits interoperability between the products of different vendors. The use of Switched Virtual Connections (SVC) would entail that paths are defined dynamically, this would increase the scope for interoperability between different solutions.

The interoperability standards for voice and multimedia over IP are defined by ITU H.323. These include endpoint negotiation and the format of the information but not issues such as encoding and security. Also, given the fact that the definitions as given in H.323 can be interpreted in more than one way, a guarantee of interoperability between the products provided by different vendors can not be given. Efforts are underway to provide interoperability between the gateways and gatekeepers provided by different vendors, this is being done by creating an interoperability profile using H.323 and H.225 Annex G standards.

In the absence of standards for these networking technologies, the interworking solutions of the near future will be proprietary. This entails that the users be aware of the technological aspects and that interoperability issues be made transparent to the users. Situations in which an interworking of technologies is desirable include corporate networks that run on Frame Relay and need to communicate with a remote location, their problem can be solved by implementing VoIP, without the need to install a Frame Relay infrastructure. The ability to use multiple voice technology over the same platform also means that migration to another technology need not mean a loss of investment. A new product being developed by RAD will facilitate VoFR - VoIP signaling conversion. This interworking between Frame Relay and IP should be advantageous as the ubiquity of IP services increases.

September 09, 2005

eBay may acquire Skype VoIP

Previously there were rumours that Yahoo and News Corp planned to acquire Skype, the leading VoIP provider. Now it is said that eBay is interested in acquiring Skype VoIP Messenger. Skype is valued around $3 billion. According to Wall Street Jounal news, eBay and Skype reached an important stage of the talks.

The reason behind eBay's interest in Skype is that eBay-owned PayPal is one of the preferred ways of buying Skype Out credits. Over the past few years, eBay has become a direct competitor with Yahoo and Google in the commercial search and auction market. As Skype is associated with Chinese market with its VoIP service, eBay has taken interest in it to gain strong foothold in the Chinese market. searchenginejournal.com reports:

Some locales the big three are competing highly in are Europe, Japan and China. Skype has also been involved in the Chinese market, and is accepted internationally as the method of making Voice Over Internet Protocol calls, which makes it even more enticing to eBay.

Read More: eBay In Acquisition Talks With Skype VOIP

Security Concern of VoIP

VoIP is gaining ground in the consumer market. More and more companies are implementing VoIP to provide a better technology to consumers at a cheaper rate. However, they are a little bit concerned over the security systems. This apprehension about the security system has forced them to tread cautiously.

But such fears are unfounded. Over the past few years, many business establishments have applied VoIP in their operations. Companies used VoIP for trunks, where security was easier. Network reliability and mobility are the biggest concern for the VoIP providers. computerworld.com reports:

Telecom service provider Primus Canada Inc. in Toronto, secures VoIP communications the same way it protects its Internet service. Primus offers VoIP services -- dubbed TalkBroadband -- to Canadian businesses and consumers.

Read More: Spam may be a future threat to VoIP

Carrying voice over frame relay, IP, and ATM - Part 1

Voice can be transferred over frame relay, IP, and ATM. The rate of growth of data networks has been greater than that of voice traffic, primarily due to the increased availability of broadband. This has led to the phenomenon of voice being sent more regularly on data networks. Voice over Frame Relay, VoFR, has been developed from the Frame Relays that were developed in the 1990's but were not suitable for carrying voice.

The growth of the Internet fuelled the demand for networks that would enable carriage of voice data in an inexpensive manner. Frame Relay, IP, and ATM are examples of packet networks designed for carrying voice and data, thereby satisfying the market need for a universal and cheap convergent technology. The next step is to improve the integration standards for the ubiquitous delivery of voice over Frame Relay, IP, and ATM.

Frame Relay, IP and ATM differ from PSTN, which is a circuit switching technology and has been specifically designed to carry voice. ATM differs from Frame Relay and IP in the sense that it breaks the data into small cells, this helps to speed up the data switching process in the network. Cell switching technologies can perform dynamic allocation of bandwidth depending upon their activities, this is known as statistical multiplexing. There is no reservation of bandwidth for a particular use, at any given time a bandwidth may be allotted as per the requirements of the network. As opposed to to cell switching networks, in a traditional circuit switching network, a line is dedicated to a call for the duration of the call and even when the call is on hold. This has the advantage of a consistent voice transmission but the bandwidth utilization is inefficient as the dedicated line cannot be used for other data transmissions even in the absence of voice transmissions.   

The packet switching networks were originally developed to handle traffic that moved in bursts. This means that packet switching networks are by default not as efficient as circuit switching networks as far as handling voice is concerned. The shared nature of a packet switching network implies that the delay in voice packet transmissions is erratic and not as less as desired for achieving a satisfactory quality of voice. Voice transmission should ideally be an accurate representation of the speaker's tone, inflection, etc. Delay in the delivery of the voice packets can lead to a communication gap, which may be further accentuated by packets being dropped along the network path.

Delay in the network can be reduced by increasing the bandwidth but it is an expensive alternative and the benefits of a shared bandwidth cannot be availed. Ideally, traffic congestion should be managed at the customer's end and at the backbone by prioritizing the flow of traffic. This had led to the development of smart access equipment for implementing procedures that help in reducing packet loss due to network congestion.

Voice over Frame Relay (VoFR): Frame Relay finds application in data networks in companies as it offers flexible bandwidth, easy accessibility, and a mature technology that supports a variety of traffic. Frame Relay offers the advantage of a predictable performance, it runs on the principle of Permanent Virtual Connections (PVCs) and is well suited for star configurations and closed user groups. Voice Frame Relay access devices (VFRADs) connect the router, SNA controller, and the PBX to the Frame Relay Network. This helps in achieving the integration of voice into the data network. MAXcess is a VFRAD that helps to surmount the difficulties in transmitting voice data over the Frame Relay without having to increase the bandwidth. It does so by employing the techniques described below:

Prioritization: VFRADs mark applications as per their reactions to delay. Voice and other time-sensitive data like SNA is given higher priority. Since voice transmissions are short and do not require much bandwidth, they do not have a detrimental effect on data traffic; in fact, they can be sent alongside other information like emails, graphics, etc that travels on the network. Different QoS packages are provided by the Frame Relay service providers. Clients prefer to purchase the best QoS for voice and SNA traffic; Non-real time variable frame and a slightly lower QoS, it is preferred for a LAN network and Internet connections. VFRAD can also be set to recognize traffic that can be dropped in case of network congestion. It achieves this by utilizing the Discard Eligibility bit. 

Fragmentation: VFRADs have the capability of fragmenting data packets to allow voice data higher priority in terms of transmission even if it leads to stopping the other transmissions. However, increasing fragmentation leads to reduced bandwidth efficiency due to an increase in the number of data frames. Applications such as RAD FR+ make it possible to send complete data frames and fragmenting occurs only if the voice data reaches a switch in the midst of a data transmission.   

VoIP protocols - Part 3

SAP stands for Session Announcement Protocol. It is an announcement protocol used in the advertising of multicast media by the session directory clients. It is also used in communicating the session setup information to participants. The multicast announcement has the same scope as the session that it announces, this helps in keeping the local session announcement local and in maintaining the scalability of the protocol.

A SAP listener uses the Multicast-Scope Zone Announcement Protocol for listening to the multicast scopes on a SAP address and port. Instead of IPsec authentication headers, application level security is used in facilitate interoperability between mechanisms that are used for announcing the sessions. The session can be announced by a web page, a session initiation protocol, or by email.

The SAP protocol structure includes:

  • V: A version number field, which is three bits and is set to 1.
  • R: It stands for Reserved and is set to 0.
  • T: It is the message type and can have a value of 0 or 1, where 0 is the session announcement packet and 1 is the session deletion packet.
  • A: It is the Address Type and can have a value of 0 or 1. 0 is the originating source field and contains a 32-bit IPv4 address, 1 is the originating source and contains a 128-bit IPv6 address.
  • C: It is the compressed bit and the payload is compressed if C has a value of 1.
  • E: It is the encryption bit and can have a value of 0 or 1. 0 implies that the packet is not encrypted and that the timeout must be absent and 1 implies that the payload is encrypted and the timeout field has to be present at the packet header.
  • Timeout: It is a value that gives the NTP time for timing out a session and is included when the session payload has been encrypted and in the absence of the decryption key, listeners may not realize the timing fields in the payload.
  • Payload type: It specifies the MIME content type and it elaborates on the payload format.

SDP: It stands for Session Description Protocol and elaborates on session announcement and session invitation. A session directory tool present on the Internet Multicast Backbone (Mbone) helps in advertising the conference sessions and provides the conference address and other relevant information. The SDP messages, which are UDP packets, are relayed by multicasting an announcement packet to a popular multicast address using SAP. The messages carry a SAP header and a text payload and can be sent across the World Wide Web by using email. SDP uses different transport protocols such as SAP, SIP, RTSP, HTTP, etc. Also, SDP does not support session content negotiation and media encodings. The SDP messages consist of the session name, its duration, media details, and necessary information to access the media.

SIP: SIP stands for Session Initiation Protocol and it is an application-layer protocol that provides mechanisms for end user systems and proxy servers to establish, change, and end multimedia sessions including VoIP calls. It can also be used to initiate multicast conferences. Existing sessions can be modified by adding or removing media from it. Names can be mapped and services can be redirected with the help of SIP. This enables user mobility as a user can now have a single identifier independent of their location. SIP supports the following facets of multimedia communication:

  • User Location: It helps in determining the end system to be used in the communication process.
  • User Capabilities: Media parameters are determined by these.
  • User Availability: It checks for the readiness of the receiver to participate in a communication.
  • Session Management: It helps in modifying session parameters and ending sessions.
  • Session Setup: It sets up the session parameters at the caller's and the receiver's end. 

SIP can be used as a component to develop a multimedia architecture like RTP that can be used to provide real-time data as well as feedback on QoS; the delivery of streaming media can be managed by RTSP; gateways to PSTN networks can be controlled by Megaco; and SDP can be used for providing information on the multimedia sessions. SIP can be used along with these protocols, however, its functioning is not impeded in the absence of these protocols. SIP is also used to provide security against DoS attacks, facilitate user to user and proxy to user authentication, and encryption.

In an Internet telephony session, SIP addresses are used to identify the caller and the receiver. A caller making an SIP call sends a request to the relevant server. The request may reach the receiver directly or it may lead to a number of SIP requests by the proxies. The SIP addresses can be present on web pages in the form of URLs, this helps in integrating them with applications like Click to talk.

The SIP messages can be sent using TCP and UDP, the messages are text based and use the UTF-8-encoded ISO 10646 character set. The messages are either requests or responses. The lines end with CRLF. An SIP request message consists of

  • Method: Methods include Invite, Ack, Options, Bye, etc and are carried out on the resource.
  • SIP version: The version of the SIP.
  • Request-URI: It is the SIP URL or the general Uniform Resource Identifier to which the request is addressed.

A response message header has the following format

  • Reason-phrase: It describes the status code.
  • SIP version: The version of the SIP
  • Status-code: It is an integer code that relates to the efforts to fulfill a request.

SGCP: It stands for Simple Gateway Control Protocol and it is an Internet protocol within a distributed system and is used to control telephony gateways, which are basically network elements that facilitate conversion between audio signals and data packets that are transferred over various networks. The SGCP works as a connection model and its two primary components are endpoints and connections. Call agents set up the connections that are grouped in calls. An endpoint consists of a domain name of a gateway and a local name inside the gateway.

September 08, 2005

First amendment Communications launched free VoIP services

First Amendment communications has decided to expand its business by launching free VoIP services in the market. Its innovative trial offer will be available to 100 members. The users will get a short ad at the beginning of the calls they initiate. After listening the ad, they will be able to use free VoIP service. It will allow them to call anywhere in the US and Canada.

Those interested in availing this service will need to have their own broadband connection. First Amendment Communications will provide the customers VoIP service and adapter. The new service is seen as a revolutionary step towards expanding VoIP beyond the boundaries. prweb.com reports:

Once the trials are complete customers may choose to port their own existing number. When the full service is launched First Amendment plans to offer the option to pay for International calls at very aggressive pricing as an add-on product.

Read More: First Amendment Communications Announces Free Voice Over IP (VoIP) Services

VoIP protocols - Part 2

MIME stands for Multipurpose Internet Mail Extensions and is a set of standards that redefines the format of messages to accommodate character sets for message bodies. These character sets are different from US-ASCII. The headers that define the structure of MIME messages are covered under RFC 2045. The initial set of media types is defined by RFC 2046. RFC 207 elaborates on the extensions that permit non-US-ASCII text data in Internet mail header fields. IANA registration procedures are specified by RFC 2048. MIME message formats and acknowledgements are illustrated by RFC 2049.

MIME enables an email to carry almost any type of text, image, audio, and video data. MIME employs base64 as an encoding procedure to ensure protection for non-text messages. It achieves this objective by coding non-text messages as text. Communication protocols such as HTTP also use MIME for the transmission of data. Messages are mapped in and out of a MIME format by email clients.  MIME was developed under the condition that the existing email servers would not need any changing. This is made possible by making the MIME attributes optional. It is possible for a MIME-capable client to interpret a non-MIME message by using its default values.

MIME type comprises a combination of type and subtype. The charset of a text type reveals its encoding. Internet protocols such as HTTP use the content-type header and MIME type registry. MIME enables messages to have a tree structure. MIME supports the following message types:

  • text messages of the text/plain type, this is the default value for "Content-type:".
  • text with attachments, this is of the type multipart/mixed with text and non-text parts. The MIME content-type and the filename extension indicate the type of file.
  • original attached to the reply, this is of the type multipart/mixed with the original message included as a message/rfc822 part.
  • messages sent in alternative formats such as HTML in which the messages are of the type multipart/alternative and having the content in other formats like text/html.   

RVP over IP: It is a proprietary specification developed by MCK Communications. It is used for the transfer of digital telephony sessions over packet networks. The signaling occurs through the TCP session and the voice is transferred via the UDP session. RVP over IP depends on the network configuration and the level of QoS. MCK offers proprietary PBX services and RVP over IP is used for connecting a remote client and a phone switch. When a remote caller attempts to make a connection with the PBX, a TCP session to the Extender PBXgateway is initiated by the MCK Extender. The initiation occurs from a high TCP to TCP 2698.

The devices communicate as client and server with the MCK Extender products functioning as clients. The first TCP port to begin with 1024 or higher is opened by a client that initiates the RVP over IP session. The client then sends a request to TCP 2698. Voice and network parameters make up the data packets. The voice parameter consists of a voice path, voice compression algorithm, DTMF encoding, comfort noise generator, echo cancellation, silence detection. The network parameters comprise packet size and jitter buffer. The remote MCK extender starts the UDP stream upon the successful establishment of the TCP session. The UDP stream starts from port 12288 (0x3000) up to 12544 (0x30FF).  The UDP listening port is 2698. RVP over IP reduces network traffic congestion and packet loss by employing a packetizer that uses a data packet for holding several voice samples.  The codec and packet size determine the interval at which voice is transmitted.

Fluke Networks offers VoIP Management Solution

Flukes networks announced that it has prepared a comprehensive VoIP lifecycle management solution. It will help the network managers to deploy, monitor, troubleshoot and manage VoIP networks. Fluke network's VoIP lifecycle management solution will manage the VoIP infrastructure for planning a future growth. It will act as a cost-saving formula and enhance competitive advantage through the new productivity tools.

Users expect IP phones to be more reliable than the voice-dedicated landlines. They also expect the VoIP systems to have high quality and performance standards. The network should provide a flawless performance in the ever-changing network environment. To minimize all the problems, Fluke networks has made a VoIP lifecycle management solution. primezone.com reports:

This lets IT managers verify before deployment that the infrastructure can support VoIP. It also allows the thorough examination of all system elements during deployment and management of the VoIP system proactively after deployment, including ongoing monitoring, troubleshooting, and planning for future growth.

Read More: Fluke Networks Offers VoIP Management Solution for the Entire VoIP Lifecycle

VoIP, Inc. launched Network 911 Service for VoIP calls

VoIP, Inc has announced the lunch of industry's first private network 911 service for broadband and packet communications. VoIP, Inc is a global provider of VoIP. The private network service 911 is provided by its subsidiary, VoiceOne Communications LLC. The Industry now-a-days has focused on creating quick solutions to meet FCC deadline which require all VoIP service providers to offer 911 services to the customers.

VoIP, Inc's VoiceOne network provides five entry points for IP 911 calls to enter the network through the Internet. These entry points are placed to provide the shortest possible path for a phone call originating anywhere in the US. Carriers and service providers can manage the entire 911 service feature set through the VoiceOne web portal. It also offers the XML provisioning system, which is not offered by solution providers in the 911 industry today. businesswire.com reports:

Currently VoiceOne is working with select carriers and service providers in the industry through its 911 product trials. VoiceOne plans to release the entire product at the CompTel/ASCENT Fall 2005 Convention and Expo in Orlando, Fla., Oct. 9-12, where demonstrations of the provisioning interface, reporting, etc. will be demonstrated at the VoiceOne Booth (#500).

Read More: VoIP, Inc. Launches Industry's First Private Network 911 Service for VoIP Calls

Challenge for VoIP: Security Appliance

The next-generation technology, VoIP is growing stronger day by day. More and more companies are adopting VoIP to provide the customers a better infrastructure at a cheaper rate. VoIP can add a number of features to business telecom systems. However, vendors and customers face tough security challenges to take advantage of these benefits.

According to In-Stat, the market research firm, three-fourth of the companies have implemented VoIP plan to replace their existing security appliances by 2006. The security appliance market is expected to grow stronger over the next few years. Traditional firewall technologies can complicate several aspects of VoIP. Security vendors are adding functions that address voice applications in their products. digitimes.com reports:

A recent report by In-Stat found that large and middle-scale companies show a higher percentage of concerns about VoIP security than small-scale companies. The report also states that budgets allocated for new security appliances are significantly higher in companies that have already implemented VoIP. In addition, reliability is apparently the most important criteria for the purchase of new security appliance products.

Read More: In-Stat: VoIP driving security appliance market

VoIP protocols - Part 1

The growth of VoIP can be attributed to the low cost and integrating of voice and data infrastructures. The components that make up a VoIP system include a Signaling Gateway, Call Manager, Call Agent, and Media Gateway Controller. The Gateway is capable of converting the media from one type of network into the desired format. Duplex media translations, T.120, audio, and video can be processed by the Gateway, which can also perform media conferencing and play audio/video messages. The voice signal is fragmented into frames by the digital signal processor (DSP). The voice signals are transmitted as voice packets across networks such as T.38 (ITU), MGCP (level 3, Bellcore, Cisco, Nortel), SIP (IETF), H.323 (ITU), SIGTRAN (IETF), etc. The bandwidth utilization is governed by coders as per the techniques mentioned in ITU-T recommendations such as G.723.1 and G.729. RTP/UDP/IP are the protocol stacks used for the conversation, which is initiated by the enabling of the codecs at both ends of the connection. VoIP provides services such as phone to phone, PC to phone, phone to PC, fax to e-mail, e-mail to fax, fax to fax, voice to e-mail, IP Phone, transparent CCS (TCCS), toll free number (1-800), class services, call center applications, VPN, Unified Messaging, Wireless Connectivity, IN Applications using SS7, IP PABX and soft switch implementations.

QoS: Quality of Service is an important aspect of VoIP communication, it covers facets such as delay, jitter, echo, packet loss, packets arriving out of sequence, etc. The QoS of a VoIP service is determined by the Mean Opinion Score (MOS). The quality of voice varies with the CODEC and MOS is used to evaluate the quality of voice for a given CODEC. PSQM (ITU P.861), PAMS (BT), and PESQ are algorithms that have been created to measure the QoS.

Megaco: It stands for Media Gateway Control Protocol and is designated H.248. It is used for separating call control from media conversion in a physically decomposed multimedia gateway. H.248 esentially defines the relationship between a Media Gateway (MG) and the Media Gateway Controller (MGC). Circuit-switched voice is converted to packet-based traffic by MG whereas the MGC controls the service logic of the traffic. H.248 can even support ATM networks, something that is not possible with MGCP. Moreover, the signaling systems supported by these network interfaces include ISDN, ISUP, QSIG, GSM, etc. Streams of data from outside a packet can get connected to a packet or a cell stream like the RTP protocol. H.246 provides a structure for gateways and IVRs. H.246 consists of two primary components, namely terminations and contexts. Analog telephone lines and MP3 streams are examples of terminations that are either entering or exiting the MG. The MGC can alter the properties of the terminations. By adding and removing the first and last termination, contexts can be either created or released. Megaco uses several commands to manage the terminations, contexts, events, etc. These commands include the following:

• Add: It is used to create a Context when executed on the first Termination in a Context.

• AuditCapabilities: This command lists the values for the Termination that are permitted by the Media Gateway.

• Modify: The events and signals in a termination can be modified by the Modify command.

• AuditValue: The latest statistics of a Termination can be learnt with the help of this command.

• Notify: The occurrences in the Media Gateway are made known to the Media Gateway Controller by the Media Gateway with the help of this command.

• Subtract: It is used to separate a Termination and its Context, when used on the last Termination in a Context, the command deletes the Context.

• Move: A Termination can be shifted to another Context by executing this command.

• ServiceChange: The MG informs the MGC via this command that a Termination has either been removed from service or is being reintroduced into service.

Media Gateway Control Protocol (MGCP): It is a VoIP protocol that is used for controlling telephony gateways, i.e. a Call Agent and a media gateway. The Call Agent houses the call control intelligence whereas the media gateway comprises media functions such as converting TDM voice to VoIP. It is between the Call Agent and the media gateway that the audio signals and other data packets that travel over the Internet undergo conversion.

A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks. Thus, it can be said that the MGCP is a master/slave protocol where the Call Agents give the commands and the gateways execute them. The MGCP is basically a connection type of model with the endpoints and the connections being the basic components. The endpoints can be both physical and virtual. There are two types of connections, point to point and multipoint. Data can be transmitted between two endpoints by establishing a point to point connection between them. A multipoint connection can be established by connecting the endpoints to a multiple session. In an MGCP model, the signaling layers of the H.323 standard are implemented by the Call Agent, which presents itself as a Gatekeeper. Transactions in an MGCP model consist of a command and a mandatory response. There are eight types of commands in MGCP, these are:

• CreateConnection: It defines the receive capabilities of endpoints by using SDP.

• ModifyConnection: Similar to CreateConnection, it modifies the properties of a connection.

• AuditEndpoint: It reveals the status of an endpoint.

• Notify: The media gateway controller is informed upon the incidence of an event.

• DeleteConnection: It ends a connection and provides relevant statistics about the connection.

• NotificationRequest: The media gateway receives requests to despatch notifications on the incidence of particular events in an endpoint.

• AuditConnection: It details the parameters relevant to a connection.

• RestartinProgress: It is used to indicate whether an endpoint is either in or out of service.

September 07, 2005

H.323 protocols and their applications - Part 3

H.261: It is basically a video coding standard that is used for transporting via the RTP with any of the protocols that support RTP. H.261 supports CIF and QCIF video frames that have luma resolutions of 176 x 144 and 88 x 72, respectively. An H.261 header consists of SBIT, EBIT, I, V, GOBN, MBAP, QUANT, HMVD, VMVD.

H.263: It defines the payload format for the H.263 bitstream in RTP. The RTP packet can use any of the three modes for the payload header of H.263. Fragmentation at the Group of Block (GOB) boundaries is supported by mode A, which is the shortest payload header. Fragmentation at the Macroblock (MB) boundaries is supported by the long payload headers, i.e. modes B and C. The H.263 payload header, whose size depends upon the modes, succeeds the RTP fixed header and the H.263 header is followed by the H.263 compressed bitstream. Payload header size is 4, 8, and 12 bytes for modes A, B, and C, respectively.

RAS: It stands for Registration, Admission and Status (RAS). It is a protocol that is used to carry out signaling functions such as registration, admission, disengagement actions, etc between the endpoints and the gatekeeper. A Request in Progress (RIP) message is used by an endpoint or a gatekeeper that is unable to respond to a request inside the timeout limits. This allows the receiving endpoint/gatekeeper to reset its timeout timer. Timeouts and retrys are particularly significant as the reliability of the RAS message channel is not very high. Important RAS messages include RegistrationRequest (RRQ), AdmissionRequest (ARQ), BandwidthRequest (BRQ), DisengageRequest (DRQ), InfoRequest (IR), etc.

RTCP: It stands for RTP control protocol and it monitors the QoS of an IPv6 RTP connection. It depends upon the multiplexing of data and the control packets. It works together with RTP in the delivery of multimedia data and provides an out-of-band control information. The RTCP header consists of the Version, P (Padding), Reception report Count, Length, etc.

RTP: It is a standardized packet format for the transmission of audio and video over multicast and unicast network services. RTP helps in payload-type identification, delivery monitoring, time stamping, and sequence numbering. The data transfer over RTP is facilitated by RTCP that enables the overseeing of data in such a manner so as to facilitate the same over multicast networks. RTP does not cover address resource reservation. RTP and RTCP function independently of the transport and network layers.

T.38: It is an Internet Fax Protocol. It deals with the transfer of fax documents in real-time over an IP network by using either TCP or UDP as per the service environment. The TCP/UDP payload carries the T.38 data. T.38 can manage fax as well as voice data over a single network. T.38 allows the use of H.323 in the same way as it is used in VoIP.

T.120: It is a family of protocols that cover the services for multilayer protocols, MCU, etc. It promotes greater operating powers that are not possible with H.231 and H.243. The Multipoint Communication Service Protocol (MCS) is covered under T.125. The procedures included in this protocol include the exchange of MCS data between two parallel MCS providers, exchange of MCS primitives between MCS providers and users. A single MAP connection or one or more transport connection make up an MCS connection.

Skype is to enter into a VoIP Joint Venture in China

VoIP major Skype Technologies is planning to expand its VoIP market in China. It is entering into a joint venture with Chinese wireless Internet company Tom Online Inc. The new joint venture company will develop and distribute a simplified Chinese version of Skype's VoIP software. Also, it will introduce premium services for the Internet users and service companies in China.

The companies will work together to integrate Skype's PC-to-PC VoIP service with Tom's more than 70 million wireless users. The companies also plan to explore Tom's mobile and Skype's Internet experience to develop new communication features for wireless Internet platforms. They joined hands for the first time in November 2004 to develop a simplified version of Skype. Skype already has established its stronghold in Chinese market, as china is one of its three op markets. cbronline.com reports:

China is home to the world's largest mobile phone market with more than 360 million subscribers. Internet usage in the county is expected to swell to about 154 million in 2007.

Read More: Skype to create VoIP joint venture in China

NETGEAR selected Centillium to produce VoIP Platform

NETGEAR has selected Centillium's newest Atlanta(TM) system-on-chip (SoC) solutions to produce its latest VoIP platform for the consumers and small-scale business market. Centillium Inc is a leading provider of broadband access solutions. Its Atlanta series includes four unique devices and easy-to-use application software. It will enable equipment manufacturers to save development time and preserve software investments across multiple platforms.

NETGEAR is committed to provide its customers the most advanced solutions. Atlanta's high performance routing and excellent voice quality will be proved beneficial for the customers. The Atlanta series are built on Centillium's Voice Services Platform (VSP). It is to be noted that VSP is an award-winning platform. home.businesswire.com reports:

The Atlanta family's innovative application software is field-proven to meet the demanding characteristics of carrier-class networks, and has been designed to be interoperable with the most widely deployed soft switches and leading VoIP service provider networks. Most importantly, its unique set of software features offers turnkey functionality for "plug-and-play" installation and service activation, allowing equipment manufacturers to start providing complete VoIP product solutions within as little as three months.

Read More: NETGEAR Selects Centillium to Speed Next-Generation Consumer VoIP Platform to Market

VoIP has taken a giant leap with PC-to-Phone Service

Internet Telephony service has reached a significant phase. More and more companies are offering VoIP technology to their subscribers. The major companies who have taken the initiative of bringing a revolution in the wireless market are Vonage, Skype, Google, Microsoft and i2Telecom. They offer cheap long-distance calls and advanced features to their subscribers at the same time. This was made possible with the arrival of VoIP technology.

PC-to-Phone service is a extended feature of VoIP. Using this service, users can make and receive calls from regular phones on their PCs provided they have a broadband connection. PC-to-Phone VoIP calls are less expensive than the conventional phone calls. startribune.com reports:

One advantage of such services is the ability to make calls through an Internet-connected laptop when cellular service is unreliable. Many people also prefer the convenience of talking while working on a PC; the services can operate while you are doing other tasks on the computer.

Read More: VoIP goes step further with PC-to-phone service

DU@LPhone: offers both the VoIP and landline together

Another VoIP phone was launched in the growing VoIP cellular market. The new DU@Lphone is seen as a revolutionary step in the VoIP technology. It has pitched itself as the world's first cordless two-in-phone. It has the facility to offer both the VoIP and landline service simultaneously.

Users can use the DU@Lphone to make calls by a PC using Skype service. Also they can attach it to the existing landline and use it as a conventional handset. The phone connects to a spare USB socket. Whie the computer is switched on, the users can speak to other Skype customers from anywhere in the house up to a range of 50m from the base station. pocket-lint.co.uk. reports:

For ordinary telephone calls you simply dial the number and press the other green button. Different ring-tones can be set up to distinguish between internet and ordinary telephones and a call log (up to 30 entries) records who has phoned either via the internet or ordinary telephone line along with the date and time of the call.

Read More: DU@Lphone offers VoIP and landline phone in one

September 06, 2005

H.323 protocols and their applications - Part 2

H.450 is a set of standards similar to the QSIG standards that detail services for ISDN networks. H.450 defines the Supplementary Services for the H.323 protocol. These include Call Transfer and Call Diversion. The H.450 protocol works with the H.225 protocol and its messages do not have a header.

H.450.1: It relates to the procedures for controlling the supplementary services between H.323 applications. It elaborates a signaling protocol that is common to all H.323 supplementary services and is based on the generic functional protocol for Private Integrated Services Network (PISN).   

H.450.2: It is based on H.450.1 and is a Call Transfer supplementary service in H.323 networks. This protocol enables a user to transfer his call with another user to a third user without having to establish a call with the third user.

H.450.3: It is based on H.450.1 and is a Call Diversion supplementary service in H.323 networks. Call Forwarding Unconditional (SS-CFU), Call Forwarding Busy (SS-CFB), Call Forwarding No Reply (SS-CFNR), and Call Deflection (SS-CD) are the services that are included in this protocol. These services are applicable during call establishment and enable the diversion of a call to another specified endpoint, which may be a voicemail or a cell phone number. The reason for the diversion, such as unavailable or busy, is specified by H.450.3 to the destination endpoint, thereby allowing it to respond accordingly.

H.450.4: It is based on H.450.1 and is a Call Hold (SS-HOLD) supplementary service in H.323 networks. It enables the interruption and re-establishment of communication between users. SS-HOLD is applicable to the audio as well as video data streams. The caller can put the receiver on hold to carry out other activities and at the other end, the other person can too initiate another call without disturbing the held call, if he so wishes.

H.450.5: It is based on H.450.1 and is a Call Park (SS-PARK) supplementary service in H.323 networks. It enables a user (Parking User) to park a call (parked to endpoint); it results in the parking endpoint achieving idle status. The Parked User experiences filler music, video, or images while he is parked. A user can pick up a parked call or an alerting call using Call Pickup. Every authorized user of the H.323 network can avail this supplementary service irrespective of the gatekeeper zone.

H.450.6: Call waiting (SS-CW) operates when an endpoint is busy with another call or another application, such as emails. The caller is made aware of the endpoint being busy and has the option of either ending the calling or leaving a message waiting callback. The potential receiver can accept, reject, or ignore the call. SS-CW occurs when all other options such as active, waiting, etc have been exhausted.

H.450.7: It deals with Message Waiting Indications that may be voicemail, fax, teletex, etc.  It is a general purpose mechanism in which a Message Center notifies the Served User at whose end A Message Waiting lamp lights up. Message related information such as the type of message, subject, and relevance can also be highlighted. Automated message retrieval and a callback request are possible in an H.323 environment.

H.450.8: It deals with Calling Party Name Presentation in which the receiver gets to see the name of the caller. The calling endpoint or gatekeeper provides the name of the caller for gatekeeper routed calls. The gateway obtains the name of the calling party from the switched circuit network and passes it to the packet network.

H.450.9: It elaborates on the Completion of Calls to Busy Subscribers (SS-CCBS). SS-CCBS notifies a caller if the receiver is busy, the receiver's endpoint can monitor activity at its end and inform the caller when it is free, upon receiving the information, the caller's endpoint then attempts to complete the call.

H.450.10: It is a Call Offer (SS-CO) supplementary service that allows a calling user to wait till the receiver reacts to the call. The receiver can accept the call after the resources become available to him. The receiver can ignore the call offered or try to make resources available by releasing or placing on hold other calls.

H.450.11: A served user can interrupt an established call by invoking Call Intrusion (SS-CI). A Call Intrusion results in the third user either being held, invited to a conference call, or force-released.  The options available with a Call Intrusion depend upon the level of authorization with the served user. 

H.450.12: It deals with the ANF-CMN service that allows for the exchange of Common Information, such as Feature Identifiers, Feature Values and Feature Controls. This information can serve as a foundation for the indications to the local user or for filtering requests. ANF-CMN endpoints can receive solicited and unsolicited services that can be offered as a combination.



Infozech allied with BroadSoft to offer VoIP Solution

Infozec software Inc. has integrated its offerings with BoradSoft's BroadWorks platform. Infozec is a leading provider of telecom billing and settlement solutions. The integration will allow the two companies to provide an infrastructure for a VoIP service provider in Mexico, Telco. Infozec's offerings have the ability to introduce new and innovative services in real time.

Infozec's solution for Telco consists of various modules. These include mediation, customer management, order entry, credit control and accounting. I will allow its subscribers to make online payments and access other Web self care options. The integration of BroadWorks and and Infozech's Billing and Settlement server will satisfy both the prepaid and post-paid billing needs. businesswire.com reports:

The BroadWorks VoIP application platform predominantly uses an open client interface; by virtue of which, service providers and solution vendors can seamlessly integrate their solutions and deliver best-of-breed services without the need for large customizations. BroadSoft's interoperable platform, coupled with Infozech's highly reliable, scalable, platform agnostic and secure products, give users the flexibility to add new and innovative services rapidly and easily.


Read More: Infozech Offers an Integrated Solution for VoIP with BroadSoft

3com launches new VoIP Applications

3Com has launched a new set of VoIP applications. Using these applications, users will be able to gain access to to a company's converged communication network. The new applications are designed to increase employee productivity. At the same time, they will reduce the costs and build stronger customer interaction irrespective of the location of the employees. The applications are targeted at the branch and mobile workers because employees can make call using IP telephone system from anywhere.

Mobile and remote users can also redirect an office call to a mobile phone, hotel room or any other office. Further, they will be able to use corporate IP telephony applications like IP messaging; IP conferencing and IP contact centres. These features will help to enhance the mobility. informationweek.com reports:

3Com and Ingate Systems are addressing an important need for enterprises to provide an open standards-based telecommuting solution which improves business processes, Zeus Kerravala, vice president of enterprise infrastructure for The Yankee Group said in a statement. It is imperative today that enterprises find a way to help their associates who telecommute and travel a great deal to remain productive while accessing their communication applications in a secure, converged fashion.

Read More: 3COM Rolls Out New VoIP Applications

Nortel brought VoIP to the Rural North American Market

Nortel announced the successful deployment of its next-generation, SIP-enabled DMS-10 platform. It is seen as a step to bring VoIP into the rural North American market. This solution is designed to expand revenue opportunities for the rural service providers. The rural service providers now will be able to offer services enabled by Session Initiation Protocol (SIP) such as VoIP at a low cost.

Nortel's rural VoIP solution gives DMS-10 subscriber the option to use traditional or VoIP primary phone service. It also allows user to add the cost-effective VoIP secondary phone service. Subscribers can also take the advantage of VoIP mobility capabilities. This will allow them use their phones while travelling provided they have broadband access. lightreading.com reports:

Nortel's DMS-10 platform allows independent service providers and rural market carriers to seamlessly advance their networks to a new, cost-effective packet infrastructure at their own pace without expensive upgrades to the network. With the new SIP-enabled enhancements, service providers now have the capability to offer new revenue generating services such as primary or secondary line service over any broadband facility, mobility options with IP phones and clients, and service bundles such as VoIP coupled with the subscriber's existing broadband or long distance service.

Read More: Nortel Deploys Rural VOIP

H.323 protocols and their applications - Part 1

H.323 is a protocol that covers the broadcast of real-time voice, video, and data over IP-based networks. H.323 is applicable to multipoint-multimedia transmissions and provides a range of services that find use in various businesses. The streams of media move along RTP/RTCP, where RTP is the carrier for the media and RTCP carries the status. H.323 is an umbrella protocol from the International Telecommunications Union (ITU). H.323 also ensures the congruence of mobile multimedia applications and different services. It comprises several protocols as explained below.

DVB: Digital Video Broadcasting (DVB) systems are supported by CATV infrastructures. DVB uses satellite, cable, and terrestrial means for the purpose of broadcasting. DVB standards were created in 1993 in Europe with the objective of unifying the framework for all delivery systems.

H.225: It is a standard used to establish a call over a Registration and it encompasses narrow-band visual telephone services as per the recommendations of the H.200/AV.120-Series. H.225 deals with managing audio and video information on a packet based network in order to facilitate services in an H.323 environment. 

H.225 Annex G:  It supplements the H.225.0 RAS protocol in fulfilling the needs of communication between administrative domains. It details the process of address resolution and access authorization necessary for completing calls between administrative domains. The H.225 Annex G is required because of the amount of equipment that exists in an H.323 network. It does not require a particular system architecture inside the administrative domain and it supports different call models like gatekeeper routed and direct endpoint.

H.225E: It deals with the implementation of UDP and TCP based protocols by using a packetization method, a signaling framework, and wire-protocol. It also specifies the profile for transmitting H.225.0 messages. It uses the security measures available under IP-SEC and H.235 and its design facilitates its use in engineering networks.

H.235: It is a security protocol for the H.3xx series. It provides authentication and integration for H.323 based systems, it enables the identification of an individual and not the application. H.235 messages are encrypted in the same way as those in ASN.1. H.235 provides point-to-point and multipoint conferencing for all terminals where H.245 is used as a control protocol.

H.323(SET): It elaborates on the standards for Simple Endpoint Types (SET). SET devices are meant for a single purpose only, they constitute a large number of H.323 capable end systems. These devices enable audio calls with other H.323 endpoints while using only a small portion of the H.323 specifications. SETs need not necessarily be PC-based, they can be relatively inexpensive applications such as the telephone.

H.245: It describes the line transmission of non-telephone signals. The features described include the sending and receiving  properties and the desired modes at the receiving end as well as Control and Indication. H.245 messages can be divided into request, response, command, and indication messages. The message sets include Terminal capability messages, Logical channel signaling messages, Request Mode messages, Round Trip Delay messages, etc.

An analysis of the promise of wireless broadband

The growth of broadband has been driven by factors such as favorable regulations, competition, and the integration of the Internet into our daily activities. Australia, which has been slow to jump onto the broadband bandwagon is set to experience a boost in broadband usage that is going to see its users jump from 25,000 in 2004 to 287,000 by 2008. Wireless broadband can serve as a single carrier for voice, video, and data and can do so without the laying down of fiber or coaxial wires that stretch for thousands of kilometers. The technology has special appeal for people that live in remote areas, such communities can experience the convenience of connectivity without having to spend on a wired broadband infrastructure, all they have to do is to have an access point that covers their area. The growth of wireless broadband requires a collaborative effort between the users and the providers. Users, residential as well as enterprise, have to view an investment in broadband as critical to long-term networking expansion. Wireless broadband promises improved delivery and better standards of service. The growth of wireless can be stimulated by developing softwares, applications, and services that push e-learning, e-governance, etc. The public sector stands to benefit as broadband will help to implement better solutions geared at promoting safer communities. Wireless broadband can help in raising the level of incident preparedness as well as incident response for emergency teams. Wireless broadband networks that facilitate high-speed voice and data transfer can keep incident response teams updated on a situation and assist in real-time decision making. Monitoring traffic situations becomes easy with wireless broadband networks that can help in regulating the flow of traffic, particularly in congested pockets. The applications of wireless broadband are limited only by the user's ingenuity. An insurance company relies upon a ubiquitous wireless network to log the real-time distances that its customers drive. This allows the company to vary the insurance amount such that it is commensurate with the risk. Remote product management implies being able to monitor and manage devices over wireless broadband. This has the potential for tremendous cost savings as well as efficient management, particularly for telecom companies. If security requirements are met satisfactorily, wireless broadband has the potential to truly enable us to live in the digital age, where connectivity is not hampered by the physical location of the infrastructure.

PABX systems are on the way out

PABX systems have been the backbone of communication in most corporates. However, the influx of new technologies such as IP networks and SIP is making IT managers evaluate their options. In many cases, PABXs are being saved from outright scrapping by the fact that they are an expensive technology and companies prefer to recover their money on them before deciding to upgrade. The rate at which the upgrades happen is also dictated by the complexity of the process. According to Mick Reegan, chief convergence officer at Nortel Asia Pacific, approximately 50% of the corporates will adopt IP based systems that will support a converged environment. The real value from implementing IP technology will be derived when applications and voice are seamlessly integrated. This may be achieved by opting for IP-enabled PABXs or IP networks only. Siemens plans to achieve this integration through its OpenScape product that will combine its data and voice networks. Yet, it appears that using an IP-enabled PABX may not be such a good idea. This is because IP-PABX infrastructures are developed on proprietary instead of open standards. Therefore , the rate of growth of IP-PABX will not be able to keep pace with the developments in telecommunications. According to a report by Gartner, voice communication in future will be dependent on applications and not infrastructures, this cannot be achieved by persisting with IP-PABXs. Instead SIP technology will be one of the important drivers influencing voice communications, this is because SIP enables applications to work with voice regardless of the equipment used. Moving to an open infrastructure enables companies to envisage new uses for their software applications and work in a more scalable and flexible setting. Integrating applications such as databases and email clients with voice call allows for improvements in processes and productivity. Robbie Kruger, development manager at Avaya solutions feels that companies will shift entirely from a legacy hardware to a server-based environment. In this regard, companies have to decide between purchasing the equipment and a hosted service. When the idea of hosting, also known as IP centrix was first mooted, it was greeted with considerable enthusiasm. It was predicted that by 2008, approximately 25% of the companies would opt for IP centrix in order to cut costs as they would not be required to install any voice switching equipment at their space. However, security concerns have resulted in a mixed response from companies. An alternative is a grouping of hosted systems that manage voice with in-house systems that manage databases. This is facilitated by the fact that the systems are being developed on open standards and therefore different systems can be linked to meet a company's specific requirements. The next step in wireless VoIP telephony appears to be the fixed-to-mobile convergence, which should eliminate the need for a landline phone. A handset can be configured through a wireless network when in the office and can be switched to a roaming network when the employee is on the move.   Another option is to install a system that routes all calls over a mobile network. Ericsson has installed such a system for a software firm HP, in Sweden, which has resulted in savings of almost 40% for the company because of reduction in call costs and by being able to reduce the number of PABXs in the company. Mobile carriers that offer this "one -phone" service allow free calls between users in a company. Moreover, PABX functionalities like auto call-backs and conferencing are available on these phones.

September 05, 2005

Wireless VoIP and savings in the office

St Agnes Healthcare, in Baltimore USA, invested in portable phones manufactured by Vocera and reported improved productivity and dollar savings. The hospital unit secretary, nurses, and nurse technicians reported saving 1650, 1146, and 626 hours of work per year, respectively. This translated into savings in excess of $ 74,000 per year for each Vocera device. 802.11 wireless connections are used to transmit voice as data packets that are passed through a speech recognition device and then transferred to the destination by routers. Vocera is just one of the many wireless communication devices, others include Spectralink Netlink, Hitachi IP-5000 VoIP Wireless Phone. VoWLAN looks set to supplant digital enhanced cordless communications (DECT), which is currently the preferred choice in warehouses, retail stores, etc. If a company shifts to VoIP as a part of its PABX upgrade, it does not have to retain DECT, which requires a different kind of management. Instead, VoWLAN devices, which work with commodity WLAN base stations already being installed by companies to maintain data networks, are a better choice. In the absence of a standard QoS structure, the quality of calls in a WLAN may vary with certain factors like vendor service and type of equipment. WLANs have design constraints that do not allow roaming between access points as routing between different subnets is not possible. This implies that the user must stay in the area where he is getting a clear reception, if he steps out of this area, he may lose the call. Avaya and WLAN company Proxim have come together to produce WLANs that enable roaming in the subnet. For a company to truly leverage the benefits of VoIP, its employees should be able to communicate using VoIP from wherever they are and should not have to fall back on another phone. This objective will largely be achieved when more and more cell phone manufacturers incorporate 802.11b for voice calls into their instruments. In order to sell, a communications device should switch automatically between GSM and VoIP networks. It will be easier to introduce VoIP in a company if the employees can get the benefits of GSM and VoIP combined in their mobiles. A survey by research organization Vanson Bourne that was performed in six countries concluded that converged devices have a huge market. For wireless VoIP to grow and find acceptance in the corporate world, GSM phones that are mobile compatible need to be developed. This calls for a partnership between VoIP and GSM specialists; for example, the partnership between Avaya and Motorola. I-Mate, which is based in Dubai has released handsets that have a Skype client, this feature enables users to make free calls to other Skype compatible mobiles from wireless hotspots. Till date, VoWLAN handsets worth US $ 45 million have been sold. According to Infonetics, the technology will achieve very good market penetration by 2009. This may perhaps not be great news for equipment manufacturing companies that depend on telecommunication carriers for business as VoIP enabled mobile phones will affect the business of these telecom carriers. If mobiles can be equipped with an authentication and encryption method that does not tax an instrument's computational and battery limitations then it would be possible to use VoIP capable mobiles for receiving calls on a network from anywhere in the world. Authentication is made possible by using a subscriber identity module (SIM) card for a challenge - response. This helps to establish the identity of the VoIP caller. In the absence of such an arrangement, it will be difficult for companies to permit an unsecured VoIP call to cross the firewalls and access the internal network.

The planning required for implementing a wireless LAN service

Voice over wireless LAN (VoWLAN) is finding increased acceptance across vertical markets in the healthcare, retail, and warehousing industry. Deploying VoWLAN presents its own set of challenges, distinct from those presented by wired VoIP requirements. Low latency with minimum jitter, smooth hand-offs, coverage, and mobility are some of the requirements that need to be fulfilled. In a WLAN setup, the bandwidth is lower as compared to wired networks and there are several devices that compete for the same bandwidth. Consequently, radio frequency (RF) has to be used intelligently in order to minimize latency. This can be achieved by implementing a mechanism that regulates traffic and prioritizes voice traffic over data traffic, thereby ensuring QoS. Wireless multimedia extensions (WME) and Wi-Fi scheduled multimedia (WSM) are standards compliant protocols for prioritizing voice traffic. The prioritization has to work with IP-based prioritizations like DiffServ code point and 802.1p. Since wireless phones have to fulfill cost, mobility, and power consumption criteria, they do not have a very robust security and authentication system. In fact, a robust authentication system can lead to increased latency and hamper seamless roaming while shifting from one access point to another. By working in a virtual network, wireless phones can work in a secure domain without connecting to a voice gateway. VoWLANs work with 802.11b that offers a bandwidth of 11 Mbps; however, to utilize the bandwidth in optimum fashion it is important that the LAN be designed correctly. Factors such as location, risk of network failure, operational loads, etc affect the performance of the radios in a WLAN. RF management and load balancing should be used to control the interference with other radios in the vicinity and to handle the calls in busy areas more effectively. VoWLAN requires sufficient coverage in order to avoid dropped calls and therefore the installations should be used only for voice traffic and not for storing data. RF management should be able to plug any coverage gaps in case of AP malfunctions or redundant WLAN resources should be provided. VoWLANs should enable seamless connectivity to enable people to use their cell phones as well. A Layer 2 (VLAN) and a Layer 3 (IP) infrastructure support roaming. Roaming with Layer 2 is easy to execute but restricts the users whereas a Layer 3 service is non-restrictive but is not very easy to implement. Since VoWLAN roaming is effected with an IP infrastructure, a Layer 3 roaming is preferred. The performance of a VoWLAN with respect to roaming can be measured in terms of the scalability of roaming, amount of jitter at an access point coupled with a different controller, etc.  Finally, it can be concluded that the success of VoWLAN depends upon the architecture of the WLAN and factors such as mobility, security, and coverage.

Know your wireless standards

802.11a, 802.11b, 802.11g, and Bluetooth are wireless standards that SOHOs and SMBs should know about as these may influence the type of WLAN equipment they purchase and the services that their vendor provides. 802.11 was the first WLAN standard that was created by IEEE in 1997, it supported a bandwidth of 2 Mbps only. It was followed be 802.11b in 1999. 802.11b supports a bandwidth of 11 Mbps and uses a radio frequency of 2.4 GHz, which is an unregulated frequency and hence affected by interference and disturbance from applications like ovens and cordless phones. 802.11b is very low cost and its signals have a good range and suffer from minimum obstruction. However, it is slow and cannot support multiple users at the same time. 802.11a was created at around the same time as 802.11b and caters to the business market. It can work with a bandwidth of up to 54 Mbps and regulated signals of up to 5 GHz. The higher frequency reduces the range of 802.11b and the signals cannot penetrate obstructions easily. As the two standards work on different frequencies, they are not mutually compatible but can be implemented alongside each other. 802.11a provides a very high maximum speed and multiple users can work on it simultaneously. 802.11g is a combination of 802.11a and 802.11b. It supports a bandwidth of 54 Mbps and a frequency of 2.4 GHz. 802.11g network adapters are compatible with 802.11b network adapters. It offers the advantage of a very high maximum speed and a high signal range. An alternative to WLAN is Bluetooth, which has the advantage of being a low-cost technology but is unable to support a high bandwidth. It works with a bandwidth of 1 Mbps and at a range of 10 meters. Therefore, even though it can network cell phones and PCs, it does not offer much value for WLAN networking. 

September 04, 2005

The DVG-G1402SL wireless router

The DVG-G1402SL is a broadband wireless VoIP router that has been introduced by D Link and Lingo. It is aimed at fulfilling the requirements of homes and SMBs. The DVG-G1402SL router can be connected with an analog phone to make VoIP calls. It is an 802.11g router that allows Internet connectivity to wireless clients. It has a 4-port switch that can connect up to 4 computers.

Vonage offers wireless connectivity

New Jersey based Vonage, which has more than 800,000 subscribers and is one of the leading providers of VoIP telephony, has signed a marketing agreement with TowerStream, a company that provides high-speed Internet services using WiMax technology. WiMax, also known as the 802.16 standard for wireless broadband, delivers speeds of up to 75 megabits per second over a distance of 30 miles. WiMax offers speeds that are more than 20 times those of wired broadband. Given that Vonage offers cheaper rates than traditional telephony, its deal with TowerStream could impact the traditional telephony and broadband services.

Funkwerk launches new SIP VoIP Phones

Funkwerk Enterprise Communications (FEC) is launching new SIP VoIP phones. SIP stands for Session Initiation Protocol. It will help users to switch over to VoIP with better quality and advance features. The new 'elmeg IP290' SIP VoIP phone has two Ethernet ports, for the network and PC. It can be connected directly to an Ethernet network.

There are a number of UK SIP service providers like SIPCall, RadiusIP, Sipgate or CallUK. Using one of the SIP service providers, users can send and receive calls from other VoIP, digital or analogue phones. The elmeg IP290 SIP VoIP phone is less expensive and has advanced features. telephonyworld.com reports:

The stylish elmeg IP290 phone features a graphical display, robust keypad and height adjustable base, and unlike other VoIP phones it has many of the features provided by conventional telephones and networks. These include programmable keys, call holding, blocking and transfer along with call back and forwarding.

Read More: FUNKWERK SHIPS NEW SIP VOIP PHONE WITH FAMILIAR TELEPHONE FEATURES

Meru Partners with Juniper to offer VoIP

Meru Network made a partnership with Juniper Network to offer VoIP products and services. Juniper's advance IP and security solutions will combine with Meru's wireless LAN portfolio. Their joint venture will provide a unique and high-class solution for the growing VoIP market. The partnership will usher a new era by bundling Juniper's IP routing and security services with Meru's Air Traffic Control technology for WLANs in order to enable IP service for mobile users.

Meru strongly believes in providing a complete solution to its customers which will satisfy their needs. The partnership with Juniper has given it the opportunity in achieving the goal. Not only its existing customers will benefit from the new venture, it eyes in wooing a large number of new customers to use its network. The long experience of Juniper in IP routing will definitely help Meru to broaden its network. tmcnet.com reports:

The companies' news release noted that "network security policies will have common enforcement, regardless of whether users are accessing the network via wireless or wireline." These features would be key for companies wanting to develop and market WLAN phones, dual-mode handsets and WLAN-enabled PDAs.

Read More: Meru, Juniper Deliver Wireless VoIP

Adomo is all set to adopt VoIP

Communications solutions firm Adomo is all set to embrace VoIP technologies. It is designing a voice messaging system that integrates with Microsoft Exchange and Active Directory. The Adomo Voice Messaging combines a list of telephony components, including VoIP. According to Adomo sources, corporate migration to VoIP is under way. Adomo is building the perfect solution for enterprises, which will enable them to save costs. Most of the enterprises now-a-days are adopting VoIP network and they will find Adomo's solution helpful.

Using Adomo Voice Messaging for Exchange, voice messages can be accessed from a phone as well as from their outlook inboxes. They can also be accessed from wireless devices. internetnews.com reports:

Andrew Feit, vice president of marketing for Adomo, says corporate migration to VoIP is well under way. However, he added, numerous challenges facing organizations must be addressed by firms, such as Adomo, in order to ease the transition.

Read More: Adomo Gives Voice to Exchange

Four important voice codecs

G.723.1 is a standards-based voice codec for video conferencing and telephony over standard telephone lines. Real-time encoding and decoding is achieved with G.723.1. G.723.1 is integral to the video conferencing standards for H.323 and H.324. It operates at a frequency of 8 KHz and a bandwidth of 6.4 Kbit / s. G.726 can work at 16, 24, 32, and 40 Kbps. G.726 converts the signals to the compressed form as per the bit rate selected. It works on the principle of Adaptive Differential Pulse Code Modulation (ADPCM). iLBC supports voice communication over IP, it is a free codec that operates at a bit rate of 13.33 kbit / s for an encoding frame length of 30 ms.  The degradation of speech quality with iLBC is not abrupt. It operates at a frequency of 8 KHz and a bandwidth of 15.2 Kbit / s.  Speex has been created for speech and is an open-source patent-free software, which intends to popularize voice applications by offering an alternative to the costly voice codecs. Speex is available under the BSD license. It works at a frequency of 8 KHz and a bandwidth of 32 Kbit / s.

JAJAH is serious competition for Skype

A free new software known as JAJAH enables users to make audio and video calls from PC to PC, to SIP-enabled phones, smartphones, and mobile phones. JAJAH apparently provides better audio results and uses less bandwidth than Skype. In fact, JAJAH does not require broadband connectivity, it does not slow down other programs, and does not take up a lot of disk space. The reason for this is that JAJAH's proprietary codec requires less than half the bandwidth that other codecs require. Its codec requires 4 Kbytes / s and hence the software can run even on a dial-up connection. JAJAH also supports the following codecs, Skype - bandwidth of 112 Kbit / s; G.711 - bandwidth of 64 Kbit / s; GSM - bandwidth of 32 Kbit / s; G.729 - bandwidth of 8 Kbit / s. It offers features such as live text chat with mobile phones, real-time text message translation, call forwarding, conference calls, searching Skype users and adding them to a contact list on JAJAH, end-to-end encryption, voice mailbox, etc. JAJAH offers calling to landlines and mobiles at very low rates. JAJAH is a P2P networking software that uses a proprietary protocol and also supports protocols such as SIP, H.323, POTS, and IAX2.  Calls to standard telephones in the US, Netherlands, Sweden, France, Italy, etc are at or below 2.5 cents per minute. Calls to Brazil, Greece, Israel, etc are at or below 1.7 cents per minute. These rates are inclusive of taxes. Currently, only a JAJAH version for Windows is available, the video quality is not very good and the frame rate is not high, adding contacts is a little tedious as both the nickname and email address of the desired contact has to be added, navigating by using the various tabs is a trifle tedious.

Skype signs deal with E Plus

Skype has sent a strong signal that it intends to diversify beyond providing PC-based VoIP solutions by signing up with E-Plus, which is Germany's third largest mobile network. The deal will allow the 10 million users of E-Plus to make free calls using Skype on the E-Plus global network. Skype is also collaborating with mobile handset developers like Motorola to come up with instruments that support Skype. Skype CEO Niklas Zennstrom has stated that the company intends to increase the number of Skype-enabled devices, such as Wi-Fi mobile phones and set top boxes, so that in the future its technology is an integral part of applications that are developed. vSKYPE is a new software that allows video conferencing through the Skype network. It is a result of Skype allowing access to developers to its application program interface (API).  By tying up with wireless ISPs such as Boingo and Livedoor, Skype plans to develop Internet telephony with the help of WiFi.

September 03, 2005

Verizon's Big Step for the transformation of the Cellular VoIP Industry

Verizon is planning to conduct trials of CDMA2000 1xEV-DO Revision A technology next year. It will be major milestone in the transformation of the wireless industry. It will eventually turn all cellular conversation into VoIP calls.

This technology is called Rev A in short. It has incorporated several advance features compared to its previous version. The improvements have extended broadband IP links carrying both the data and voice to the handset. That is beneficial for both the service providers and users. It lets service providers to lower their capital and operating expenses for network equipments. voip-magazine.com reports:

Of course, some such savings are possible simply by going to IP gear in the transport network, regardless of the air interface, observes Ruchi Prasad, director of global CDMA product marketing at Nortel. The more exciting benefits come from the fact that the handset itself becomes a native IP device, which makes it possible to extend all the cool features of IP telephony to cellular users.

Read More: Verizon's Big Step Towards Cellular VoIP


Damovo and N.E.T. joined hands to launch VoIP environment in Brazil

Damovo, a global communications solutions and service provider joined hands with Network Equipment Technologies (N.E.T.) to announce SHOUT VoIP Migration Appliance (VMA) in Brazil. The solutions offered by the SHOUT VMA enable companies to migrate their networks to VoIP environments. For over 12 years Damovo and N.E.T. worked together to provided services in different sectors like Finance, Energy and others. Their partnership is now set to conquer VoIP market.

The SHOUT VMA will allow the enterprises to change their legacy communications systems to a converged IP solution. It will also enable the Call Centres to adopt the new generation VoIP applications. SHOUT is capable in operating in a multi-vendor networking environment. Damovo and N.E.T. are also planning to expand their new VoIP application to the Government and Defence establishments in the country. businesswire.com reports:

SHOUT's multi-vendor, multi-protocol support, combined with its ability to support open standards such as SIP enables enterprises to reduce implementation and capital costs when integrating their existing networks with VoIP technologies. In addition, SHOUT also provides advanced voice encryption features for secure communications, a requirement in some vertical segments.

Read More: Damovo and N.E.T. Launch Platform for VoIP Environment

Cable One selected Sigma system's VoIP solution

Cable provider Cable One selected Sigma Systems as its OSS vendor to deliver new PacketCable telephony to its subscribers. Cable One currently has over 700,000 subscribers. The reason behind choosing the Sigma Systems based on the fact that it has better infrastructure and long experience in providing OSS installations for VoIP. By choosing Sigma, Cable One has scored a point over its rival cable operators. Sigma's OSS solutions for VoIP and broadband IP services are the most trusted ones in the cable industry. Sigma is confident that its VoIP Service Package and other features will enable Cable One to easily deploy new next-generation services to subscribers.

Cable One will use Sigma's Service Management Platform (SMP) and VoIP Service Package to provide services like dial tone, voice mail, long distance calls etc. Sigma has already proved itself in the field on VoIP deployment across the globe. tmcnet.com reports:

The role of OSS service management is essential as we deploy VoIP services, scale operations, and look to grow revenues for our subscriber base, said Steve Fox, Cable One's vice president of digital services and technology, in a prepared statement. Using Sigma's solutions, we gain operational efficiencies by significantly reducing the manual work and processes associated with service order management and service provisioning for complex VoIP services.

Read More: Cable One Chooses Sigma Systems' OSS VoIP Solution

Wi-Fi VoIP will not make the Telephone Operators redundant; it will enhance the Technology

There is a suspicion all over that VoIP will mark the beginning of the end of telephone operators. But British Telecom (BT) tried to dismiss such fears as unfounded. Though the VoIP is entering into the Cellular market as an alternative to 3G, it is too early to write off the fixed-line operators. VoIP is not a disruptive option. In fact, if we ignore VoIP completely and stay with traditional PSTN, it will not be conducive to the growth of telephony.

BT is now building a VoIP network to offer broadband connections. Instead of charging per minute, it will charge the users on a monthly basis. Some VoIP over Wi-Fi calls will be free of charge to the users. ferret.com reports:

BT has 7,500 Wi-Fi hot-spots in the UK, which will be used as a complementary service to 3G telephony. Dual mode handsets, expected to become available either next year or in 2007, will be able to switch between GSM and Wi-Fi.

Read More: Wi-Fi VoIP will not spell end for operators, BT says

VoIP in Indonesia

Indonesia is all set to embrace VoIP in its wireless market. MOBIF Bhd, a maker of Internet-based systems, has signed up a deal with a distributor in Indonesia, Jogja Medianet that is into sales and purchases of surveillance and telephony products. Jogja Medianet is planning to buy MOBIF's Internet surveillance systems and both the broadband and dial-up VoIP boxes. According to CEO of MOBIF, Indonesia is a big market as many of its citizens working or studying abroad. Thus, VoIP can prove to be a helpful tool to attract more consumers.

The biggest benefit of VoIP is it's cost-saving factor. It can save a lot of money for the consumers and at the same time provide a high quality of communication mode. Mobif's technology allows the calls to be made with fixed-line or mobile phones, by using a VoIP box connected to the Internet. nst.com reports:

The founder of Jogja Medianet, Dani Sudrajat, said his targeted markets for Mobif’s products are educational institutions and the Indonesian Government in Jogjakarta and West Java. Parents can monitor their children in schools using the surveillance systems or make cheaper calls to their children studying abroad through VoIP, he said.

Read More: Mobif confident of mart potential in Indonesia

Microsoft-Google tussle over the supremacy in VoIP market

In a significant development to the growing competition in the VoIP market, Microsoft and Google are set to engage in a tussle over the dominance on VoIP market. The battleground is set for both the companies to woo the large consumer market. Google already announced to launch Google Talk, a VoIP service. Google Talk is a Windows application for instant messaging and PC-to-PC voice calls.

Microsoft joined the fray by acquiring Teleo Inc, a VoIP service provider. Using the VoIP services offered by Teleo, it is planning to expand the features on its MSN Messenger service. MSN Messenger already has the features of calling from PC using a speaker or Microphone. Now Teleo's technologies will enhance its capabilities of conducting voice applications. informationweek.com reports:

Also this week, Google released Google Desktop 2, which updates its desktop search software with new personalization features. In July, Microsoft launched a test version of a search engine that lets users search PCs and the Web with a single tool.

Read More: VoIP Marks Latest Microsoft-Google Battleground

VoIP is set to conquer the Cellular Market

The growth of public Wi-Fi, Instant Messaging and new enhanced cellular handsets has paved the way for the entrance of VoIP into 3G cellular world. Yahoo Messenger and MSN Messengers have dominated the IM services on the PCs and handsets for a long time. However, with the VoIP gaining momentum they are lagging behind VoIP providers like Skype Technologies in providing excellent features of talks. Recently Skype announced a deal with Handset Manufacturer Motorola to provide its VoIP features on certain Motorola 3G handsets. However, Motorola has not confirmed when it is going to launch the phones featuring Skype client.

The deal between Motorola and Skype has put significant pressure on their rivals to adopt VoIP phone service. Yahoo and MSN have been slower in developing VoIP as their IM clients for mobile phones do not have VoIP service. To bolster its VoIP service, Yahoo recently acquired Dialpad communications Inc. neasia.nikkeibp.com reports:

It's clear that Yahoo is pursuing the international VoIP telephone audience that Skype, with more than 43 million users globally, has come to dominate. The larger question is if, or when, Yahoo will be able to roll-out a VoIP client in mobile phones operating on the cellular network of the largest cellular carrier in the US, Verizon Wireless.

Read More: VoIP Gains Traction in Cellular Market

Service providers, consider these facts before starting a VoIP service

The target markets for VoIP service providers include residentials, SOHO, SMB, and enterprises. Each of these markets has distinct requirements that a service provider can try to address in order to acquire long-term customers. Residentials look for cheaper local / long distance calls and PC / mobile integration. SOHOs require service assurance and PC / mobile integration. SMBs need corporate features at reduced expenses. Enterprises want consistency of service across their offices and the flexibility to outsource their network management. In the US, 50 million homes and up to 55% of the adult population has access to broadband Internet, almost all the SMBs have high-speed Internet connectivity, and VoIP line shipments to the businesses are as high as TDM line shipments . These factors indicate a rapid acceptance of VoIP and underscore the need to select the target market segment intelligently. For service providers, capital and operating costs are major factors in deciding whether to build or buy the VoIP solutions. A managed solution does not offer a great deal of network control and options in product packaging; however, it does enable quicker deployment of systems that have been proven. IP PBXs have several drawbacks that VoIP service providers can capitalize on. IP PBXs are fast becoming technologically obsolete and sometimes require periodic hardware upgrades, the user interface is not very convenient and requires at least a fortnight's training, IP PBXs cannot interface smoothly with equipment from other vendors. Selecting the right distribution channel for VoIP can be a trifle tricky as indirect channels need to learn more about this new service and direct distribution is not a very cost-effective solution. By offering a cost saving in the range of 10% - 30%, service providers can hope to make inroads into the TDM market, in fact SMBs have evinced a strong interest in VoIP upon being offered cost savings of 15%. A service provider should try and leverage the expertise of the network and software vendors that he may work with. They should back their service with dependable SLAs and assist in testing and providing media gateways, customer premise equipment (CPE), etc so that the service provider is up to date with the latest applications available in the market. Before graduating from the pilot stage and going live, a service provider needs to ensure redundancy of the production systems; media servers, network connectivity and other supporting equipment should be tested; User Acceptance Testing (UAT) and troubleshooting procedures should be in place. Since VoIP is unlike a traditional data service and it is different from a conventional voice service, it has its own peculiar engineering issues such as Local Number Portability (LNP); 911, e911, and 800 services; corporate CLEC procedures; call flows; etc. These can be tackled with the expertise available in-house or by partnering vendors.

Managing your migration from a WAN to a MPLS

Multi-Protocol Label Switching (MPLS) offers several advantages, these cover cost, scalability, and reliability issues. MPLS enables better network management and tracking down of malware and unwanted traffic and server / client and application related issues can be pinpointed more efficiently. Before initiating a VPN / MPLS migration, a company needs to look into the bandwidth consumption of the applications, the number of ports that the firewalls will manage, the effect of the transition on the responsiveness of the network, and tools to measure the performance of MPLS. Network Physics offers a solution for better network application management. It offers baseline performance measurement, real-time enterprise-wide visibility of data centers and NOCs, boosts bandwidth by eliminating rogue traffic and removing viruses, facilitates capacity planning and tackling response time issues.

Sony Ericsson Launched VoIP-enabled Bluetooth Headset

In a new dimension to the VoIP gadgets, sony Ericsson launched a new Bluetooth Headset which is VoIP enabled. This amazing product is known as HBH-608. The new headset is compatible with the Sony VAIO BX series notebooks. It's bluetooth wireless technology allows the users to talk hands-free.

This technology connects the notebook with the headset. This helps users to talk vial VoIP. Wiring and other telephone equipments become redundant for the users who use this headset. HBH-608's wireless connection enables users to roam around within a distance of 10 meters. tmcnet.com reports:

According to the companies' news announcement, "the acoustic construction of speaker and microphone in the HBH-608 ensures high quality sound, whether it is used with a computer or a mobile phone." The device is small, lightweight, and it extends talk time to up to 10 hours.

Read More: VoIP Gadget Spotlight: Sony Ericsson Intros Bluetooth Headset

Katrina has created a debate: the need of VoIP technology during disasters

Technologies such as VoIP can be a major helpful tool during the natural calamities and disasters. The recent devastation caused by Hurricane Katrina proved this. In the wake of Katrina, the focus has been shifted to the need of a better technology for data-recovery procedures and backup systems. One area that is most likely to receive more attention aftermath Katrina is VoIP.

IP phones may be able to prove as a cost-saving factor for the customers during such catastrophes. Risk of communication disruptions is very low in case of IP phones. Using the VoIP system, data and information can be accessed anytime anywhere which is not possible in other normal mode of communications in the situations like Katrina. It will work even if some of the services on the network are not available. informationweek.com reports:

Finally, the whole concept of distributed data centers that can automatically pick up workloads from each other should drive a lot more interest as we develop the next generation of automated server provisioning tools. These tools allow for the configuration and provisioning of a server in a matter of minutes, rather than the days it might take to do the same thing manually.

Read More: Opinion: Katrina Sharpens Focus On Emerging Business-Continuity Technologies

Cost savings with broadband service provider MASERGY

Dallas based Masergy communications offers its customers a feature-laden service at a cost that is approximately 30% lower than its competitors. It offers the following features - virtual private LAN service that is compatible with any access technology that the customer may be using, a secure and guaranteed bandwidth, video conferencing at 30 fps and zero "tiling", IP voice VPN capability, a private IP VPN having security that is equivalent to that provided by a Layer 2 service, and the facility of combining public and private networks on to a single circuit. Masergy uses Multi-Protocol Label Switching (MPLS) to regulate the flow of data packets across it global Ethernet WAN service. The company provides network reporting and management tools that are accessible through a web server.

Assessing VoIP quality in an enterprise-wide deployment

Assessing the readiness of a network prior to a deployment can help in minimizing training costs and obtaining better results with the pilot deployments. Vivinet Assessor is one such software that helps in predicting a network's readiness for VoIP. VoIP demands unique network requirements that may entail network upgrades; an assessment report pinpoints the areas that need to be upgraded. The Vivinet Assessor uses a Mean Opinion Score (MOS), which is derived from the G.107E model, in order to provide an accurate feedback on end-user perception of the call quality. 

September 02, 2005

Cost benefits of using a GSM VoIP gateway

When a call is routed from a fixed line to a mobile phone, it passes through the Mobile Telephone Switching Office (MTSO). This is an expense for the PSTN operators, which they pass on to the fixed line users. Given the current scenario in which up to 50% of the telephone costs of a company are fixed line to mobile calls (F2M), this is a sizeable amount for many companies.  Two VoIP GSM gateways, namely 2N VoiceBlue Lite and 2N VoiceBlue Enterprise offer the advantages of IP telephony over traditional PBX's; the main advantage being a reduction in the F2M and M2F call costs. The VoiceBlue Lite gateway functions with the SIP and RTP protocols that allow it to communicate smoothly with the IP PBXs. The VoiceBlue Enterprise gateway is compatible with both SIP and H.323 and can work in both environments simultaneously. VoiceBlue Enterprise has an SIP proxy built into it, which allows it to be used as an IP PBX. This makes it ideal for remote locations and temporary setups where landlines may be either difficult or expensive to deploy. VoIP GSMs offer cost savings of up to 50% on F2M calls and an attractive ROI in less than six months. The Least Call Routing (LCR) facility on VoiceBlue gateways allows savings even on national calls.

X-Pro TAPI to integrate VoIP with Outlook

Global IP Telecommunications introduced the X-Pro TAPI, a Telephony API-compliant software telephone which can be integrated with Microsoft Outlook. The X-Pro TAPI is the evolved version of X-Pro Softphone. It can be noted that X-Pro Softphone is the award winning SIP telephone from the market leader Xten Networks. It supports the real TAPI, which helps to link professional CRM software with VoIP. It also providers a Caller ID feature to the users.

The interesting feature of this VoIP Softphone is that it can be integrated to Microsoft's Outlook. With the X-Pro TAPI feature, MS Outlook users will be able to any member on the Outlook contact list. The product will create new opportunities for the VoIP market. tmcnet.com reports:

The new product brings professional functionality to VoIP users, which has previously been reserved for large installations. The development was difficult, extremely time consuming and could only be accomplished through excellent expert knowledge and skills. says Michael Best, director of Global IP Telecommunications, Ltd. in Schoeffengrund, Germany.

Read More: X-Pro TAPI Integrates VoIP with Outlook

German Operator, E-Plus tied up with Skype to provide VoIP service

German operator E-Plus announced a tie-up with Skype to offer VoIP service. This move will enable customers to avail VoIP over wireless calls at a flat rate. E-Plus has been placed third among the Telecom Operators in Germany. It always eyed to expand its network throughout the country by providing the latest features to the customers. It scored a point over the rivals by choosing VoIP software developer Skype to develop its infrastructure.

E-Plus is currently owned by Dutch telco KPN. It will bundle the Skype software with its flat rate data subscription for US$50 per month. This service is expected to be launched in the coming October. telecoms.com reports:

E-Plus said that it would only offer Skype software for its VoIP service and that the offering would also be available over its UMTS network. Skype said that it already has more than 2.8 million fixed line based users throughout Germany and added it is working with a number of mobile handset and headset manufacturers, including Motorola, to offer a range of Skype-ready devices.

Read More: E-Plus signs up Skype for VoIP offering

September 01, 2005

Hanaro Telecom, first Service Provider in Korea to offer VoIP services

In a major announcement made by Lucent Technologies, Hanaro Telecom will deploy Lucent's VoIP solution to offer advanced services to the customers. Hanaro Telecom is Korea's second largest wireless and broadband service provider. It will be the first Korean company to provide VoIP service. Lucent will enable Hanaro to provide existing services like call forwarding and conferencing, as well as innovative IP-based services. The advanced web-based features will allow the customers to manage their voice services via their computers.

Hanaro has a strong communication network in South Korea for local, national and international voice services. Using this network, Lucent can expand its base not only in South Korea but also in the Asia region. Recently Lucent announced a contract with Dacom to build IP networks for the Militay Mutual Aid association. prnewswire.com reports:

The solution, which includes the Lucent Feature Server 3000, also can allow Hanaro's enterprise customers to consolidate their voice, data and Internet services onto one network that connects to the public network, simplifying the enterprise's network operations. Lucent Worldwide Services will provide professional services including engineering and installation for the deployment.

Read More: Hanaro Telecom Selects Lucent Technologies VoIP Solution to Offer Advanced Services to Enterprises

VoIP 2.0, the future of VoIP

VoIP has started a new revolution in the history of Internet Telephony. With more software companies embracing VoIP, it is recreating itself with the advent of the new technology. It is believed that future of VoIP is VoIP 2.0, which will focus on services instead of cut-rate pricing. Though companies like Google, Skype and Microsoft are offering new services now, the day is not far when every company will be forced to charge for this service after adding new features and technologies.

VoIP 2.0 is the latest version of VoIP which will allow the users to advantage of more flexibility, customization and powerful features. It is being planned to introduce more advance features rather than the simple conferencing feature. It will be the next phase of VoIP. VoIP 2.0 is the theme of Internet Telephony Conference & Expo to be held this October in Los Angeles. Using VoIP 2.0 may prove to be more productive, efficient and profitable. tmcnet.com reports:

Google, Microsoft and Skype are software companies providing primarily a free service. Where do they have to go but to add more features and new services?

Read More: VoIP 2.0 Gets Closer

Microsoft enters VoIP market by acquiring Teleo

In a significant development, software major, Microsoft acquired Teleo, a provider of VoIP. Teleo is a major VoIP provider, which offers this technology through software and web applications. By acquiring Teleo, Microsoft has taken a giant leap forward to establish itself in the growing VoIP market. Microsoft is expected to combine both the existing technology and MSN applications. Though Microsoft already offered VoIP through its MSN messenger, integration of Teleo with Microsoft may usher a new era in VoIP market. Till now Micosoft has not discosed the amount for which it bought Teleo.

Teleo, founded in 2003 has its headquarter in San Francisco. It is a privately held company. It has established itself as one of the major players in the VoIP market. digitalmediaasia.com reports:

Founded in 2003 and headquartered in San Francisco, Teleo is a privately held company whose initial planned service offering, also called Teleo, was designed to allow customers to use their PC to make phone calls to mobile phones, regular phones or other PCs. Through its integration with Microsoft Outlook and Microsoft Internet Explorer, the Teleo service was designed to facilitate click-to-call dialling of any telephone number that appears on-screen, for example through a website or via search results or e-mail.

Read More: Microsoft enters VoIP market

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