Mitigating the Latency Induced Delay in IP Telephony Through an Enhanced De-Jitter Buffer

IP telephony or voice over IP (VoIP) at present is promising a shining future for voice services. There are several technical aspects which make the technology attractive; on the other hand, few technical loopholes and shortcomings make user’s experience less than optimal and also bring forth significant security issues. This paper offers a technical dissection of the quality of service (QoS) of VoIP. “Signaling” part of VoIP has been discussed based on the Session Initiation Protocol (SIP) along with propositions to tackle problem like jitter that often causes latency in communication. To address the issue of jittering, an alteration in the working mechanism of de-jitter buffer has been put forward where it is shown that addition of few extra variables within the de-jitter buffer to synchronize the packet arrival and release timing can certainly improve the user experience. Reducing the latency is of prime importance to voice data services as it directly affects the acceptance trend of VoIP among mass consumers. The scale of improvement has also been compared to that of a normal jitter buffer as well as a detailed illustration has been provided on Session Initiation Protocol (SIP), a key component of the overall system that makes thing happen. The proposed modification in the de-jitter buffer has been illustrated along with positive results. It shows a one-third improvement in the average latency, resulting into twice as better performance and nearly halved latency.