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.

通过无线边缘网络进行语音传输是一种通过 802.11 边缘网络传输语音呼叫的传统且经济高效的技术。在此类服务中，会话发起协议 (SIP) 负责发起会话。研究表明，控制SIP会话的呼叫控制算法也可以保证音频数据传输过程中的媒体质量。尽管在无线接入点进行决策并改变媒体接入子层(MAC)的参数，但现有算法复杂且不能充分保证服务质量和系统资源的有效利用。因此，本研究的目的是提出一种新算法，使 SIP 服务器能够在简洁的时间内决定接受或拒绝来电。该算法在不改变MAC子层设置的情况下，考虑了网络的动态参数，如节点位置、信道繁忙率、信道实时流量百分比等。每个系统的位置取决于传输呼叫的接入点，对提供所需服务的方式有很大影响。在真实网络测试台上实施这种位置感知方法表明，在服务质量方面，如响应时间、呼叫持续时间、丢失率和实时延迟等参数衡量的服务质量明显优于最近提出的方法数据包。每个系统的位置取决于传输呼叫的接入点，对提供所需服务的方式有很大影响。在真实网络测试台上实施这种位置感知方法表明，在服务质量方面，如响应时间、呼叫持续时间、丢失率和实时延迟等参数衡量的服务质量明显优于最近提出的方法数据包。每个系统的位置取决于传输呼叫的接入点，对提供所需服务的方式有很大影响。在真实网络测试台上实施这种位置感知方法表明，在服务质量方面，如响应时间、呼叫持续时间、丢失率和实时延迟等参数衡量的服务质量明显优于最近提出的方法数据包。