Internet has had offered plethora of services availed by the users either free of cost or at unimaginable price. Voice over Internet Protocol (VoIP) is also one of such boons provided by the net. It is a technique for transferring Voice (analog signals) over the Internet (packet switched based) using Internet Protocol (IP). Voice transmission through a non-circuit switched network rather than a dedicated path brings many impairments in its implementation like Delay, Jitter, Spiky delay, Echo and Packet loss. These factors contribute a lot to the quality of service (QoS) and quality of experience (QoE) delivered to the user. The playout buffer at the receivers end apart from buffering of the packets, performs the reordering and setting the playout time of voice packets that reach the destination through different routes. Researches have been made that aims in modifying the palyout buffer algorithms so as to convey the best QoS or QoE to the user. This paper discusses the fact that rather than considering, calculating and altering the buffer algorithms against the factors contributing to VoIP impairments, it must be adjusted according to the bandwidth and components used for VoIP implementation.