Just as in classical telephony, the voice signal is sent to the phone microphone handset in the form of an analog signal. An analog to digital converter transforms this signal into a digital one which is then encoded according to an audio format. There are different encoders for compression of a conversation with varying degrees of quality. Depending on the type of compression, some loss of voice signal information occurs, which, however, is mostly subjective and irrelevant.

Voice Data, after the compression, is transferred via the network. For this continuous data stream of compressed voice signal, the conversation is divided into small data packets, before they are sent to the network. Then, these packets arrive at the specific destination network “nodes”, so-called routers, which direct the IP-packets to their final point, maybe via different paths.

The voice packets, before they are sent to the network line, they are first temporarily stored in a memory location called the buffer. At the destination end, voice packets pass through a digital to analog converter so that to be transformed back to human speech. For example, the voice packets delivery can be compared with normal mailing post, which addresses, mails and delivers individual packets of data.

To transfer data using VoIP we currently use the so-called Internet protocol version 4 (IPv4). The next version – IPv6 – was specified in the mid 90-ies, but the need to expand its use has not been taken seriously until the last few years. Unfortunately, the IP protocol works on the principle of “Best Effort” and thus it does not guarantee 100% packet delivery. This is the main reason that we do not always have good quality in VoIP. The newest IP Protocol version 6 offers the so-called “quality of service” under which the voice data transmission will have better quality.

In order to establish a connection between a VoIP telephone system and the traditional PSTN telephone network, we need to use the so-called Gateways. These Gateways are connected to the IP Data network as well as with the PSTN telephone network and transmit requests in both directions. In this case, IP-packets from one side are converted into digital voice stream on the PSTN side.

Since everything now runs over IP (our computer data, voice, video etc) we can have the integration of different types of data on to a single IP network. This concept is usually called “network convergence”. The meaning of convergence is that we have one common network (the IP Network) which transmits all kinds of information – voice, data, video, text and images.

The Cisco Unified Communications Manager Express (CUCME) is the new brand name given by Cisco to the older Call Manager Express (CME) system. The concept is the same however: IP Telephony software running on Cisco routers. Therefore, the CUCME is a normal Cisco router (models supported are 1800, 2800, 2900, 3800, 3900 series) with a special IP Telephony software (call manager software) installed on the router’s flash memory. The CUCME system serves as the call control node to facilitate IP Telephony communications in a small to medium size Enterprise.

Usually there is a single CUCME system in each LAN network, with several IP phones connected on the LAN switches. An enterprise with several sites connected over a private IP WAN network can establish full IP voice communications between sites by configuring H323 communication between each CUCME router. A simple example with a two-node topology is shown below.

CME-A node has local IP phones with numbering 500x and a WAN IP address of 1.1.1.1. On the other site, CME-B has local IP phones with numbering 600x and a WAN IP address of 2.2.2.2. By establishing H323 voip communication over the WAN (between 1.1.1.1 and 2.2.2.2) we can have full IP telephony conversations between the IP phones of both sites.

CAUTION: Because the actual VoIP RTP traffic communication between site A and site B will be running from one IP phone to another IP phone, there must be full IP routing established between the IP phone subnets.
The CUCME configuration to establish H323 between the two sites is shown below:

CME-A

CME-A#show running-config
dial-peer voice 6000 voip
destination-pattern 60..
session target ipv4:2.2.2.2
dtmf-relay h245-alphanumeric
codec g729r8

CME-B

CME-B#show running-config
dial-peer voice 5000 voip
destination-pattern 50..
session target ipv4:1.1.1.1
dtmf-relay h245-alphanumeric
codec g729r8

The dial-peer configuration on CME-A tells the system that in order to reach the destination pattern 60xx the session will be established with IP address 2.2.2.2 (i.e CME-B). The inverse applies for CME-B.

Note: Make sure to select one of the high compression codecs ( such as g729, g723) in order to save bandwidth for voice calls over the WAN network. Each VoIP conversation using a high compression codec (g729, g723) will use significantly less bandwidth compared with the traditional G711 codec.

Continuing our series of posts on IP Telephony and VoIP, here is Part 3 of the tutorial:

Is IP Telephony Implemented Easily ?

Over time, most companies have acquired the expertise to implement IP Telephony solutions, either on existing corporate networks or from scratch. The main advantage to implementing VoIP applications is that they rely on network infrastructure which can be expanded gradually, depending on the needs of the business. Additionally, complimentary applications have been matured as well, such as call management software, so that the implementation of solutions and their use becomes more straightforward.

What happens in terms of voice quality ?

Traditionally the main problem of telephony on IP networks has been the quality of the voice. Since the same network carries different data packets (documents, other voice conversations etc.) we cannot always ensure that the packets carrying the voice conversation will all get together and on time at the other end in order to carry a real-time discussion. When you transfer a document, a web page, an email etc, we don’t care so much if one packet is delayed 1-2 seconds. In voice conversation however, delay works negatively on the quality of the voice. A solution to this problem would be the usage of high-capacity lines, combined with powerful routing equipment (eg routers and large enough switches). However they cost money. A better solution is the implementation of prioritization of voice packets with respect to other data. Gradually, as the cost of equipment and services drops, the quality of VoIP will be better and better. Finally, we must not forget that using certain technologies (e.g voice compression), we can increase the efficiency of communication lines and with appropriate settings in routers we can commit certain capacity from the network for voice communication. With that, voice transmission will be conducted as much as possible in real time, without delays and distortion.

Do we need special telephone handsets ?

There are special telephone handsets designed for VoIP communication that harness the potential of this technology. Such devices are available from most international manufacturers of telephony products as well as from third party manufacturers involved in related VoIP solutions. It is worth mentioning that using special equipment we can still use normal telephone devices. Alternatively, a company may consider the option of softphones. A softphone is essentially telephony software that is installed on a laptop or desktop computer and offer all the functionality of an IP telephone without the need for a hardware telephone device. Of course, the use of softphones depends upon the existence of a computer. Although the first softphones presented had poor voice quality and a great network load, now the technology is fairly mature and operational.

Continuing our series of posts on IP Telephony and VoIP, here is Part 2 of the tutorial:

Can an IP Telephony System be connected to the public telephone network

There are special voice gateways which can connect an IP Telephony system with the public switched telephone network (PSTN) or other telephone networks. Using the voice gateway, a VoIP phone can call a legacy telephone line phone on the public telephone network and vice versa with no problems. Basically the voice gateway translates the IP packets from the IP Telephone system into TDM voice to be transmitted over the legacy PSTN network. Generally, regardless of the infrastructure that the IP Telephony system uses to carry out the conversation, ultimately it is a private telephone network, such as those implemented in corporate call centers, which is transparent to the public telephone network.

What are the benefits of IP Telephony and VoIP

The main advantages of VoIP and IP Telephony in general include:

• Single network infrastructure for data and telephony. Since the same infrastructure (communication lines and equipment) serve voice traffic and data traffic, we have significant economies of scale. Also, we achieve better management of telecommunications infrastructure.
• Maximum use of telecommunications infrastructure. The packet switched networks (e.g IP Networks) make better use of their bandwidth capacity in comparison with traditional circuit switched telephone networks since the line is not fully occupied for each call conversation therefore it can carry various data packets in addition to voice.
• Improved communication for remote workers. The use of IP telephony does not require the user to have a physical presence in the enterprise environment. If the user has an IP connection, he/she can take advantage of the features and functions of the enterprise telephone system, regardless of where the user is located.
• New services are introduced. The usage of a single infrastructure for both data and voice allows for the development of a new generation of services such as unified messaging that can contribute significantly to productivity growth.

Why companies are interested for IP Telephony

Since almost all companies have access to the Internet, they have already implemented their corporate networks over the IP protocol. Thus, they are given a first class opportunity to utilize the IP network infrastructure, which includes, in addition to the communication lines, other equipment such as routers, switches, etc. This IP network infrastructure can be used for telephony as well. Even if the IP telephony system is confined within the enterprise, the benefits are significant. When a company uses leased circuits to connect remote branches, the use of these circuits for both IP telephony and data connectivity provides substantial benefits and cost savings to the company.

Is IP Telephony the most economical solution for voice communication

Like any technology infrastructure investment, usage of VoIP and IP Telephony should be treated as a medium to long term business. According to studies, the use of packet switched networks for voice telephony is more economical than the networks that occupy the whole communication line for each conversation. And when we can serve phone calls through our corporate IP network – which in some cases is extended to different parts of the city, other cities or other countries – we certainly save money by not using the public telephone network. When routing phone calls over our own private IP network from New York to Los Angeles and the destination call is a PSTN number in Los Angeles, the call will be charged as local in Los Angeles (it will be routed from our voice gateway in Los Angeles to the PSTN). This is an example of a toll bypass cost saving. Companies should however consider the costs for the implementation of the IP telephony infrastructure, occurring in the increased bandwidth capacity to accommodate also voice traffic, in the extra equipment (e.g IP telephones), the additional software needed, etc. Overall, however, in medium to long term, telephony over IP networks has proved to be much more economical than traditional telephony solutions.