Voice and data convergence reduces costs by lowering bandwidth requirements, simplifying maintenance, reducing power consumption, centralizing management and increasing productivity. VoIP (Voice over Internet Protocol) converts analog voice signals into IP packets and distributes them across a WAN (wide area network). POTS (plain old telephone service) quality VoIP requires a finely tuned network. QoS (quality of service,) traffic shaping, compression and Byte-level cashing, fragmentation and packet aggregation can ensure high quality jitter-free voice over existing data networks.
Voice and data convergence reduces costs by combining voice and data on the same Internet connection. VoIP automatically compresses data and only uses bandwidth when sending and receiving information. This allows several phone conversations to take place while sharing the same Internet connection eliminating the need for a separate voice line.
Maintenance is simplified using VoIP. When moving someone’s desk or office it is not necessary to change any wiring. Simply plug the existing phone in to a power strip, connect the Ethernet cable and it will automatically reconnect itself to the network. There is no need for equipment such as punch tools, wire cutters or copper wires.
Power consumption is reduced by eliminating unnecessary equipment. There is no need for a patch panel, separate T1 processing equipment, talk switch or voice mail system. Eliminating this equipment reduces power consumption, saving money and reducing the impact on our environment.
Centralizing management reduces cost by condensing voice and data management. By consolidating equipment, less office space is required, and fewer employees are required to manage the system. Settings can be made either remotely or from a central location eliminating the need for a dedicated workforce.
Productivity is increased by integrating real-time communications into the business process. Unified messaging allows users to check voicemail, faxes and e-mails in priority order with a single in-box. Users can then return messages by phone, fax or email without having to take their hands off the computer. This helps to keep the employees on task and promotes greater productivity.
When implementing VoIP there are many factors that must be considered. End –to-end delay and Jitter are extremely important when determining sound quality. It is recommended that end-to-end delay be less than 150 ms for good quality voice. If the delay becomes too long the speech becomes unrecognizable.
End-to-end Delay includes propagation delay and serialization. Propagation delay is the speed of the data traveling from point A to point B. Distance and speed directly affects propagation delay. In a modern network fiber optics allow information to travel at the speed of light. Serialization is the delay caused by the devices that handle the voice data and is determined by the equipment used to transmit the signal such as a router or switch. Routers can very greatly in performance and may take longer to transmit a packet if it is performing other tasks such as encryption or compression.
Jitter is the variation between the expected packet delivery and actual time of delivery. Jitter is caused by LAN congestion, low bandwidth and the transmission of large data packets. It is important to ensure application congestion doesn’t cause jitter on your voice calls and dedicate the appropriate bandwidth for your VoIP communications. QoS, traffic shaping, fragmentation, compression and packet aggregation help to guarantee jitter-free high quality VoIP communication.
QoS can be used to ensure VoIP packets takes priority over other applications. By ensuring that the VoIP packets are placed on the line first, we can greatly reduce jitter. Data packets that are not as time sensitive can be buffered and placed on the line later giving priority to VoIP packets.
Traffic shaping can be used to guarantee performance, improve latency and increase usable bandwidth for VoIP. By limiting the amount of bandwidth used by greedy applications such as FTP and web browsing we can ensure proper amounts of bandwidth are available to VoIP applications.
Fragmentation can be used between sites to allow large non real-time data packets to be broken up into smaller packets. This prevents large packets from blocking the router interface. Web browsing HTTP packets for example can be up to 1500 bytes and take 93.75 ms to leave the router interface over a 128 Kbps link. If these packets can be broken down into smaller packets they can be transmitted faster reducing the impact of serialization delay by allowing the QoS to send the VoIP packets without having to wait for lengthily HTTP and FTP packets to be sent.
Compression and Byte-Level Cashing can also be used between sites to increase the available band width for applications by compressing data and cashing commonly used data. Although VoIP is already compressed, compressing other data can further increase line availability and reduce end-to-end delay. Cashing between sites can further reduce the transmission of data by eliminating the need to send the same files multiple times. For example, an inter office e-mail that is sent to everyone can be transferred across the internet to a satellite office once then copied to everyone without having to send the file separately to each individual.
Packet Aggregation reduces packet header overhead costs by joining small LAN packets including Citrix, RDP and VoIP together reducing fragmentation and increasing compression. A VoIP packet for example is 64 bytes. When a 32 byte header is added in order for it to cross the WAN we increase the packet size by 50%. By combining three 64 byte packets and using one 32 byte header we can reduce the number of bytes sent by 64. This helps to reduce bandwidth requirements by eliminating unnecessary data from being transferred over the Internet.
Organization using voice and data convergence can reduce costs by lowering bandwidth requirements, simplifying maintenance, reducing power consumption, centralizing management and increasing productivity. POTS quality VoIP requires a finely tuned network. QoS, traffic shaping, compression and Byte-level cashing, fragmentation and packet aggregation can all help to ensure high quality jitter-free voice over existing data networks.
