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چکیده
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Voice transmission over wireless edge networks is a conventional, cost-effective technology to transmit voice calls over 802.11 edge networks. In this type of service, Session Initiating Protocol (SIP) is responsible for initiating sessions. Research has shown that the call control algorithm that controls SIP sessions can also guarantee the quality of the medium during audio data transmission. In spite of decision-making in wireless access points and changing the parameters of the medium access sublayer (MAC), the existing algorithms are complicated and cannot adequately ensure service quality and efficient use of system resources. Therefore, the purpose of this study is to propose a new algorithm that enables a SIP server to decide to admit or reject an incoming call, in concise time. This algorithm considers the dynamic parameters of the network such as the location of node, channel busyness rate and real-time traffic percentage of the channel without changing the settings of MAC sublayer. The location of each system depending on the access point that transmits the call has a great impact on how the required service is provided. Implementation of this location-aware method on a real network testbed is indicative of remarkable superiority over the recently proposed methods in terms of service quality as measured by parameters, such as response time, call duration, loss-rate, and delay in real-time packets.
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