Requirements for VolP
The IP-based systems and networks that will be used for voice or video transmission over IP must meet several criteria to deliver a signal of the same quality that has been available through circuit-switched, digital systems and with which the users are satisfied.
Because of the real-time, interactive nature of VoIP the average transit time (i.e., latency) between source and destination is most important. As shown in Table 2.1, ideally one-way latency should be less than 100 ms. If the round trip delay on a voice conversation exceeds three-tenths of one second, then it becomes difficult to continue an intelligent dialog, as many have experienced when a telephone call was routed through a geostationary satellite.
Jitter is the variability of packet arrival times at the receiver and is generally caused by large bursts of data interfering with the real-time traffic in the network. Although modest jitter delays have not been considered serious to real-time conversation, yet significant jitter (i.e. delays approaching 60 ms) leads to unintelligible speech patterns and ruins video displays.
Packet loss corresponds to link dropouts, which is a common experience with mobile/cell phones in congested areas and also results in unsatisfactory voice or video communications.
The bandwidth, or bit rate, requirements for VoIP depend on the type of encoding/decoding (i.e., codec) technique that is used in the digital signal processor (DSP) in the end stations. A reasonably good guideline, with current technology, is to allow 20 Kbps in the network or the access link for each simultaneous voice conversation, or 200 Kbps for a small-group video conference session.
LAN Requirements
In a large office complex there may be a correspondingly large number of simultaneous voice calls on the LAN, in the busy hour of the week. This traffic volume depends on the peak telephone utilization level in the organization, which may be 25% in many offices, but rise to nearly 100% of the population in some highly sales-oriented businesses, such as a stock brokerage. Even if there are 1,000 concurrent voice calls on the LAN, this probably represents an added load of only 20 Mbps, which is insignificant on a gigabit backbone.
However, if the in-building network is to carry IP-based video traffic, as will increasingly be the case as multimedia applications are implemented, then a major LAN upgrade is much more likely to be needed. If we assume a minimum bit rate of around 200 Kbps for two-way video interaction and up to 2 Mbps for higher definition, full motion, video displays, then a number of simultaneous video sessions will put a heavy load on the LAN. In some early installations for IP-based video conferencing a separate LAN that went all the way to the main edge router for the building was dedicated to the video outlets.
WAN Requirements
An existing metropolitan area network (MAN) or WAN will certainly be more challenged by the added burden of carrying VoIP packets than a LAN. The transit and jitter delays and bit rate requirements will each have to be seriously addressed across all stages of the larger network. Generally a data network that runs on frame relay (FR) technology will not meet the latency criteria that we have specified. There may be some exceptions if the FR virtual circuit involves only two frame switches. The regular Internet is definitely not suitable for corporate VoIP traffic and only a managed network can meet the requirements.
A managed IP network is one where the service provider(s) monitor and control the QoS, outside the customer's own environment, to minimize latency and jitter, while ensuring that sufficient bandwidth is available for busy traffic. In a managed network the VoIP packets pass through the least number of nodes and the QoS is guaranteed, going beyond just "best effort" engineering.
The characteristics of the access links between the LANs and WAN must also be well defined and managed to fit within the overall criteria budget from source to destination.
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