A number of organizations are hesitant to adapt the latest communication methods and prefer to stick to the Stone Age methods of communication. They find it hard to adjust to the current ways due to technophobia and also the costs that come with investments. Communication methods are changing and with each passing day, there are new ways that are adapted. The latest introduction is the use of Android applications and iPads that enhance portable communication. It is now a reality to communicate at any time of day or night without even reporting to the office. Many workers have access to computer either through the mobile phone or through the computer. In the organization setting, many companies are adapting the use of video conferencing and free video chat to eliminate the huge transport ad accommodation costs that are incurred along the way.

There are many companies that have adapted this method and they are reaping great rewards. They find it easier to communicate with different parties they transact with. If they are looking to work with some of the big organizations, they will choose this method since most organizations use his method. They want to know who they are talking to and their presentation skills.

It is also a safe method because the parties that are involved are the only ones that will access the information. Many top companies use this method and especially the top level management board when they want to discuss a private matter in detail.

This method is also applicable to organizations that are involved in lengthy discussions and presentations and they want to inform all their parties at the same time. Video conferencing allows different people to communicate together even when they are miles apart. Technology now allows the compression of video and conversion to different video formats (a good video converter by the way is avs4you), thus video can be transferred via networks easier as it requires less bandwidth.

It is a costly method to implement but if you are not using the method on a regular basis, you can contact some of the leasing companies that hire the conferencing equipments at a fee. However, if you use this communication method on a regular basis, it will be important to buy the equipment to save on costs.

You also need to ensure that you have constant supply of internet connectivity by investing in the bandwidth or WiMAX technology. This way, communication will be applicable at any rime with no worry of interruption due to low internet connectivity.

Video conferencing is effective to organizations that are mainly dealing with long distance or international clients.

I have received some very exciting news from the CEO of Phone.com so I thought to Blog about it and give you something exciting to know about Phone.com!

Exciting News! Message from the CEO, April 1, 2011

“As the First Quarter of 2011 comes to a close, I wanted to share with you some of the excitement within Phone.com. Our company continues to grow month after month in terms of customers, functionality, and customer satisfaction.

During the past year, we introduced our Smart Phone applications, starting with the iPhone and Android and recently joined by Blackberry. Using the Phone.com Mobile Office, our customers can place and receive calls on cell phones using their Phone.com phone number! Phone.com Mobile Office works in conjunction with any Phone.com service plan.

Phone.com has also worked very hard to bridge the gap with traditional mobile phone services. For example, text messages (also called SMS messages) can be sent or received by your Phone.com account. This even works for extensions within the Virtual Office! Incoming text messages to your Phone.com number can be configured to forward to your cell phone. You can send text messages from your PC and have them appear as if they originated from the Phone.com number.

Have you visited our Promotions page recently? We are constantly providing innovative offers as incentives to new customers interested in enjoying the Phone.com experience.

We have also been very much in the public eye. In November 2010, we announced the availability of fully automated voicemail message transcriptions as part of our services, and later in that same month, announced that our company was named Communications Company of 2010 by the New Jersey Technology Council (NJTC). In February 2011, we were recognized by Internet Telephony magazine as Product of the Year! As you can see, we have been busy!

Despite all of this activity, we strive to be better and want to hear from you! Please let us know if there are new features or functions that we should consider. Our mission is to deliver superior quality of services, range of services, customer support, and overall value. Your input is not only desired, it is also essential for the accomplishment of our mission!

We are off to a great start for 2011 and with your continued support, we will all continue to grow and prosper!”

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VoIP protocols and technologies are divided into two major categories: Distributed and Centralized. The MGCP protocol is an example of Centralized VoIP protocol since it supports a centralized client / server architecture. On the other hand, H.323 and SIP protocols are categorized as Distributed since the voice distribution is based on inter-node ad hoc network.

All VoIP technologies and protocols use a common protocol for speech distribution which is RTP (Real Time Protocol) which packetizes voice traffic over IP, as well as supports multiple codecs to compress the data.

The differences lie in the way of signal transmission and in the area of service logic and mode of the call management (at the end points or on a central server). Both architectures (Distributed and Centralized) have their advantages and disadvantages. Distributed models scale well and are more flexible (robust) because they have no central node, which can lead to failure. Centralized model are more easily managed and support the traditional supplementary services (such as conferences), but may have limits on scalability which determines the capacity of the central telephony server.

A simple pulse-code modulation of speech PCM (standard ITU-T G.711) enables the transformation of speech at 64 kbit / sec of the mu-law and A-law. In both of these methods we can achieve 12-13 bits PCM quality in 8 bits using a logarithmic compression.

Another commonly used method of compression is adaptive differential pulse-code modulation (ADPCM). A typical case is the use of ADPCM coding ITU-T G.726 with a 4-bit quantization, providing a transfer speech rate of 32 kbit / sec. Unlike PCM, the 4-bit code is not the actual amplitude of the speech, rather it is the difference in the amplitude and the rate of change of amplitude, using some rather primitive linear prediction.

The new methods of compression, such as LPC, CELP, and MP-MLQ, use additional features in the waveform in both PCM and ADPCM, using knowledge of the original features of the formation of speech. Such methods are applied methods of signal processing, which compress speech by sending only simplified parametric information about the original form of sound and the vocal tract. To send this information requires less bandwidth. These methods can be combined into a common set of codecs in the source.

Currently, we have the following methods of encoding in voice over IP:

*G.711 – PCM-method at a rate of 64 kbit / sec
*G.726 – ADPCM – method of transmission at speeds 40, 32, 24 and 16 kbit / sec.
*G.728 – CELP – a method with a transfer rate of 16 kbit / sec
*G.729 – CELP – a method with a transfer rate of 8 Kbps / sec
*G.723.1 – MP.MLQ – a method with a transfer rate of 6.3 kbit / sec and CELP – a method with a transfer rate 5.3 kbit / sec.

When implementing VoIP networks we need to focus on the requirements for bandwidth and latency (delay).

Bandwidth requirements.

Bandwidth requirements are critical and are determined not only by the transmission rate of the codec used (from 3-4 to 64 Kbit / sec), but the extra load on the network, called IP headers, and other factors. Due to the presence of pauses during conversations, a technology was developed for detecting voice activity (Voice Activity Detection, VAD). With VAD, bandwidth requirements are reduced roughly in half. Thus, for example, for the G.711 codec with bandwidth of 64 kbit / sec, with the use of VAD technology the total bandwidth for a voice channel will be about 40 kbit / sec.

Delay.

Requirements for quality in voice networks determine the maximum latency of 150-200 ms. A greater delay value usually reduces the quality of conversation significantly. The greatest amount of delay (30 milliseconds) is introduced by the G.723 codec, and the lowest delay (0.75 ms) is found in G.711 codec. It should be noted that the smallest propagation delay time is introduced by channel switching networks, and the greatest propagation delay is found in packet-switched networks (IP networks) due to buffering. In connection with this fact, the VoIP technology is less attractive for voice transmission over the Internet, than VoATM and VoFR. Nevertheless, VoIP quality over the Internet is quite acceptable for a corporate network that needs maximum of 4 -6 concurrent voice channels.

Currently there is an increased interest in the implementation of triple play communications (voice, video and data) in corporate computer networks. This is because the technologies that support these solutions show their economic viability in a time when traditional telephone services reached their highest point. The introduction of IP telephony technology in the information infrastructure of the organization results in a real reduction in the cost of telephone calls. In addition, the move to IP-telephony provides a unified approach to network administration and security issues of corporate information. No need to purchase additional protection for video and voice for security since they both use the same tools as those used for the protection of the common data circulating in the corporate network.

The term “IP-telephony” means much more than just voice transmission over the IP protocol. In fact, this term denotes a wide range of multiservice networking technologies for integrated data, voice and video into existing corporate data networks. The support of voice in IP networks in the long term will replace the services of voice and video delivered through the traditional telephone network.

The main reasons for the transition to multiservice network technologies are improving the quality of customer service, cost reduction and innovation that increases the competitiveness of companies. Multiservice applications and services offered by leading manufacturers of network equipment, include:

• unified messaging
• call centers
• personal telephony
• data sharing (using tools such as NetMeeting)
• interactive and recorded video

The main technological bases for triple-play services are the protocols for voice over ATM “(VoATM),” Voice over Frame Relay “(VoFR),” Voice over IP “(VoIP), and their supporting products.

Voice in the ATM network (Voice over ATM, VoATM) is using a standard emulation of the speech channel with pulse-code modulation (AAL1) or transfer to the ATM cell with a variable bit rate (AAL2). The main advantage of voice transport in ATM networks is guaranteed quality of service (QoS). Nevertheless, VoATM inherently has excessive complexity, high cost (due to the use of optical communication channels), and lack of support from manufacturers. VoATM was commonly used as a connecting highway or transit exchange between remote PBXs.

Voice over Frame Relay (Voice over FR, VoFR) is favorably compared to Voice over ATM cost and ease of use (administration). Nevertheless, the absence of Quality of Service in Frame Relay networks with the traffic saturation that is usually found in such networks, voice traffic may experience delays and hence reduction in voice quality.

Voice over IP networks (Voice over IP, VoIP) differs from the above technologies in that the transport happen at the network level and therefore voice packets can be carried out both in WAN (including networks of ATM and Frame Relay), and in local area networks (Fast Ethernet, Gigabit Ethernet).