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VoIP (voice over Internet protocol) technology has rapidly been growing up recently, which transmits voice packets by using the user datagram protocol (UDP). VoIP quality is difficult to expect because it is hard to predict the influence of packet delay, packet lose, packet error, etc. This paper proposes an embedded call admission control (CAC) mechanism by applying real-time transfer protocol (RTP) and the real-time control protocol (RTCP) for VoIP services over hybrid fiber/coaxial (HFC) networks. The proposed CAC mechanism is evaluated by the impact of the various traffic load in cable modem termination system (CMTS), which estimates how VoIP quality satisfies the user's requirements under different constrains on cable networks. We discuss VoIP CAC mechanism for the upstream channel according to the data over cable service interface specifications (DOCSIS) version 1.1 and particularly consider G.723.1 voice packets at the transmission rate of 6.3 kbps. The performance measurement of the proposed embedded CAC mechanism is obtained by simulation experiments under various network constrains, which includes throughput, packet dropping ratio and call blocking ratio. Obviously this paper provides an efficiency and fast method for CMTS to decide how many calls can be accepted.