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VOIP SYSTEMS

VoIP Differs from Data Communications

Thursday, June 05, 2008 ≡ Mario Stocco

Ethernet and IP networks were designed to accommodate data applications that have a higher tolerance for packet loss than networks designed for voice traffic. Lost data packets are handled by Transmission Control Protocol (TCP) and are merely re-transmitted by the sender. Voice streams on a data network are packetized are sent by UDP; without any guarantee that they will reach their destination. Dropped and delayed packets during transmission results in poor sound quality. Lost packets and transmission delays are typically due to network congestion, often from poor bandwidth utilization and/or insufficient WAN link capacity.

Voice over IP telephony requires network equipment to provide identification for different types of traffic. Type of Service (ToS) distinguishes the packet type and tags it for WAN transport. Quality of Service (QoS) enables switches and routers to prioritize voice packets in order to minimize packet loss and transport delay. Network equipment purchased before 2002 typically does not have QoS capabilities. Yet, adequate traffic prioritization is critical to avoid lost or delayed packets and ensure high voice quality.

Jitter, a natural occurrence in a data network, can impact the quality of voice over ip traffic. It results from a variance in the time stamps that are placed on both the sending and receiving ends of the packet. This variance can be corrected by adding buffers to store the packets and feed them in a steady stream to the application.

Compression can be used to manage VoIP traffic across limited bandwidth (WAN) links. Packets are compressed for sending but then must be de-compressed again when they are received at the other end of the network. IT managers who wish to use compression to manage bandwidth must balance the communications impact of using less than the highest quality packets. In addition, multiple passes through compression/ decompression cycles can cause poor audio quality.

Because each network is unique in terms of its configuration/use profile, a successful migration to Voice over IP telephony requires an evaluation of the quantity and type of voice traffic it carries; monitored during peak network activity periods. A customized series of tests conducted at intervals on a live network and using a variety of parameters can capture network performance under a range of loads and conditions. By calibrating these tests to produce predictors of industry standard scores, enterprises can obtain the most relevant information to help assure a successful IP telephony deployment.

If you are in the Greater Victoria, Southern Vancouver Island area and would like a professional network assessment before you deploy a Voice over IP system, please feel free to contact me

About this blog

With nearly a decade of in-the-trenches computer telephony experience, Mario Stocco writes this weblog to articulate his thoughts on topics like VoIP, Open Source and life in general.

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