Voip Glossary

This page discusses - Voip Glossary

VoIP Bandwidth 

  1. Lines and VoIP Bandwidth Calculator
    This calculator can be used to estimate the bandwidth required to transport a given number of voice paths through an IP based network. Reverse calculations are also possible. These estimate the number of voice paths that can be transmitted though an IP network if the available bandwidth is known. Before a calculation can be performed, details of the voice compression scheme must be entered into the first two areas of the calculator. * Use the first drop down box to select the CODEC being used. CODECs convert analogue voice signals into data streams through sampling and quantisation. CODECs vary in their quality and delay characteristics and, although there is not yet an agreed standard, G.723.1 and G729A are the most common CODECs used for Internet voice transmission. * The frequency at which the voice packets are transmitted have a significant bearing on the bandwidth required. The selection of the packet duration (and therefore the packet frequency) is a compromise between bandwidth and quality. 

  2. VoIP Bandwidth Calculation
    The amount of bandwidth required to carry voice over an IP network is dependent upon a number of factors. Among the most important are: * Codec  and sample period * IP header * Transmission medium * Silence suppression The codec determines the actual amount of bandwidth that the voice data will occupy. It also determines the rate at which the voice is sampled. The IP/UDP/RTP header can generally be thought of as a fixed overhead of 40 octets per packet, though on point-to-point links RTP header compression can reduce this to 2 to 4 octets (RFC 2508). The transmission medium, such as Ethernet, will add its own headers, checksums and spacers to the packet. Finally, some codecs employ silence suppression, which can reduce the required bandwidth by as much as 50 percent.

  3. Bandwidth is Required for VoIP Phones?
    A long-standing question for potential VoIP (Voice over Internet Protocol) consumers is first of all, Bandwidth is defined as the ability to transfer data (such as a VoIP telephone call) from one point to another in a fixed amount of time. The higher the bandwidth speed you have, the more data you can send over your Broadband Internet connection. There are two types of bandwidth at your location: upload bandwidth and download bandwidth. The Upload Bandwidth is the amount of data you can send to the Internet and download bandwidth is the amount of data you can receive from the Internet. The more Internet bandwidth you have from your ISP (Internet Service Provider) the better.

  4. Algorithm cuts VoIP bandwidth requirement
    Looking to reduce edge-access bottlenecks, Effnet Inc. has developed a licensable version of the Compressed Real-Time Protocol algorithm to support the real-time delivery of voice-over-Internet Protocol. The CRTP algorithm can compress a typical IPv4 header from 40 bytes to as little as 2 bytes, allowing a T1 line to increase its call capacity over 250 percent, the company said. Header compression complements the voice payload compression typically handled by the G.7xx vocoder algorithms. However, header compression becomes increasingly important as the size of the payload shrinks," said Rich Stamm, marketing director at Effnet . CRTP compression will lower the bandwidth requirement by about 60 percent.
  5. VoIP Bandwidth Utilization and Packet Handling
    Despite long-standing forecasts that bandwidth will become essentially free and unlimited, the practical reality is that bandwidth utilization remains important for most voice over IP (VoIP) providers. Getting trunk bandwidth to points of presence (POPs) for thousands of calls can quickly become expensive. In packetized voice communication systems - VoIP, frame relay, or ATM - voice data is digitalized and lossy compressed into frames. Each frame represents the voice data for a small unit of time, typically 30 milliseconds. Frames are then transported over the network from a source to a destination. The frames are then decompressed. In a VoIP network, each voice data frame is typically encapsulated in one datagram. The Internet Protocol (IP) imposes a minimum of 20 bytes of header, containing such information as the destination IP address. The User Datagram Protocol (UDP), typically used for voice transport applications, adds another six bytes of header information. For example, a voice frame encoded with the G.723.1 voice algorithm at 6.4 kbps is 24 bytes long.
  6. VOIP Bandwidth Management Devices
    NetReality's WiseWan? technology allows organizations to prioritize mission-critical traffic, get the most out of existing bandwidth, guarantee bandwidth for voice quality and put off or eliminate costly WAN bandwidth upgrades. Often enterprise networks are stretched to the limits for WAN (Wide Area Network) bandwidth, but are still expected to deliver real-time business-critical service, such as Voice over IP (VoIP) communication. Adding WAN bandwidth with more T1, E1 or other trunk lines is one solution, but at a substantial recurring cost increase. Many times a more practical and efficient solution is to add bandwidth management that will enhance network quality of service (QoS). QoS is even more critical with real time applications like Voice over IP (VoIP). Congested network connections can cause poor voice quality that is out of the control of a device such as a MultiVOIP gateway, but with the application of QoS devices like WiseWan? from NetReality Inc. quality of VoIP service can be guaranteed.