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Performance Evaluation for VoIP with Codec, QoS, and OPNET
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- Author(s) / Creator(s)
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Voice over Internet Protocol (VoIP) is a popular topic for the networking technologies. It is a very attractive and cost effective technology that merges both data and voice networks into providing several benefits including cost savings, flexibility, advance features, and low bandwidth requirements.
However, VoIP delivers real-time voice packet across networks using Internet Protocol (IP) instead of traditional Public Switched Telephone Network (PSTN) and it is difficult to guarantee voice quality when VoIP is implemented on the real networks because voice quality is affected by several factors such as delay, jitter, packet loss, and etc. IP traffic also is naturally treated as “best-effort” and transmitted on a first-come, first-served basis. Therefore, voice codec schemes and QoS are carefully chosen to guarantee voice quality before deploying VoIP to the real networks. Codec schemes define voice compression mechanism and have different characteristics. QoS is one of network congestion managements and each queuing has different characteristics. (As cited in abstract.) -
- Date created
- 2011-04-01
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- Type of Item
- Report