In a run through the present office scenario, we have lot of single man office or mobile office. That too spread across. All the users required to access different services. Some of them are in house hosted or some of them are from internet. Companies are integrated the work flow through the devices. Here users are using different gadgets, laptop, soft phone, note book, Black Berry, I Phone, etc.

Cost Effective Communication

All the areas communication cost is an integral one. The ways of cost involved is different ways. Internet charges, telephone charges, leased line cost, ADSL cost, LAN phone cost, mobile phone cost.

Couple of Solution:-

1. Office Communicator
2. Open Source SIP based Unified Communication
3. VPN over ADSL to ADSL
4. BlackBerry

These are just high level of technologies, work each solutions depends on the size of organization, requirement, infrastructure availability, skill set. A detailed exercise required.  Out of the box, the solutions give you a major reduction in communication cost.

Smart Phones & Tablets to Work

Smart Phones & Tablets to Work

Last week I read the Toastmaster Magazine June 2011 edition. Article “Presentations TO GO!” Written by Dave Zielinski describing about couple of tools which useful for presentations and toastmaster speeches.  That is the thread I’m presenting here.

Applications which are supports smart phones & tablets. Those applications are very useful to work from smart phones & tablets. I’m listing those applications below.

• Keynote Presentation Design Software
• Documents to Go
• ThinkFree Mobile
• Prompster
• ProPrompter
• SpeechPrompter
• Dragon Dictation

• Phreaking
Phreaking is a slang term coined to describe the activity of a subculture of people who study, experiment with, or explore telecommunication systems. This is sometimes called the H/P culture (with H standing for hacking and P standing for phreaking). Through this making calls through another user network, putting cost to that person.
• Eavesdropping
Eavesdropping is how most hackers steal credentials and get important information like business data. Through eavesdropping, a third party can obtain names, password and phone numbers, allowing them to gain control over voicemail, calling plan, call forwarding and billing information.
• Vishing
Vishing is a party calling you faking a trustworthy organization (e.g. your bank) and requesting confidential and often critical information.
• DoS (Denial of Service)
DoS attacks can be carried out by flooding a target with unnecessary SIP call-signaling messages, thereby degrading the service.
• SPIT (Spamming over Internet Telephony)
SPIT attacks consist of sending a voicemail to a person, masquerading it with information from a party trustworthy to the receiver, like a bank or online paying service, making him think he is safe.
• Man-in-the-middle attacks
VoIP is particularly vulnerable to man-in-the-middle attacks, in which the attacker intercepts call-signaling SIP message traffic and masquerades as the calling party to the called party, or vice versa. Once the attacker has gained this position, he can hijack calls via a redirection server.

Fast IVR for Cisco is a powerful tool for generating and evaluating TCL scripts for Cisco VoIP-enabled gateways. Fast IVR for Cisco enables the telecom operator to execute and evaluate Cisco TCL IVR scripts locally on a workstation without the need to remove an expensive piece of hardware from production network system. Moreover, Fast IVR for Cisco tool is infused with deep experience in developing and implementing high-capacity flexible Interactive Voice Response programs and strategies – it is the most effective and innovative tool available for customer care, contact, and communications.

IVR solution is an essential component in the marketing efforts of top-tier companies in the growing VoIP space. Our IVR solutions offer telecom operators with state-of-the-art technology, so that they can provide superior customer service and very flexible products to their expanding customer base.

A SIP trunk is a service offered by an ITSP (Internet Telephony Service Provider) that permits businesses that have a PBX installed to use Voice-over-IP (VoIP) also outside the enterprise network by using the same connection as the Internet connection.

SIP trunking is a way to enjoy significant savings on your current phone bill. Using an Internet connection right from your current PBX, a SIP trunk uses SIP (Session Initiation Protocol) for a VoIP connection.

In these tougher economic times, more and more businesses, both small and large at looking at the advantages of SIP trunking and creating call treatments in their PBXs to take more chargeable calls out through these cost saving routes. And this trend is likely to substantially increase over years.

SIP trunking takes data, voice and video out over your Internet connection.

Some of the benefits of SIP trunking include:
• Much lower rates on long distance calls, International calls and in-bound toll free calls.
• The ability to establish virtual numbers in other geographic areas so that callers can use these numbers instead on more costly 800 numbers.
• Using virtual numbers from other areas to establish a “point of presence” for the company.
• Reduced costs as no TDM cards are required.
• Getting the benefits of Hosted VoIP without having to abandon existing equipment or investing in IP phones.
• The ability to easily add more calling trunks without the need for expensive digital PRI / analog station (port) cards.
• The ability to use an IAD for connecting regular analog phones, instead of having to buy more expensive equipment.
• Expansion of lines is dependent on bandwidth, which can easily be increased if needed. Additional lines can be set for compressed codecs (G729 and others) to use less bandwidth per call.

Ref: TMCnet.com

Telecommuter solution that embeds virtual private network (VPN) remote capabilities into IP telephones. Businesses can use this solution to more easily and securely extend headquarters-quality communications to employees working from any home office, temporary work site or emergency location.

The number of telecommuters and home-based workers are expected to continue rising over the next few years, according to IDC Research. By 2009, 10 million U.S. workers, or 10% of its workforce, will be telecommuting. In Western Europe, 9 million workers are expected to telecommute, representing a growth rate of 10% from 2004-2009¹.

Ref: Cisco & Avaya

SIP supports five facets of establishing and terminating multimedia communications:
• User location: determination of the end system to be used for communication;
• User availability: determination of the willingness of the called party to engage in communications;
• User capabilities: determination of the media and media parameters to be used;
• Session setup: “ringing”, establishment of session parameters at both called and calling party;
• Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.

SIP is a component that can be used with other IETF protocols to build a complete multimedia architecture, such as the Real-time Transport Protocol (RTP ) for transporting real-time data and providing QoS feedback, the Real-Time streaming protocol (RTSP ) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP ) for describing multimedia sessions. Therefore, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.

SIP provides a suite of security services, which include denial-of-service prevention, authentication (both user to user and proxy to user), integrity protection, and encryption and privacy services.

SIP works with both IPv4 and IPv6. For Internet telephony sessions, SIP works as follows:-

Callers and callers are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended caller, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their location(s) with SIP servers. SIP addresses (URL) can be embedded in Web pages and therefore can be integrated as part of powerful implementations such as Click to talk.