<![CDATA[The proliferation of Voice over Internet Protocol (VoIP) has revolutionized modern communication, offering cost-effective, feature-rich alternatives to traditional telephony. However, the inherent packet-switched nature of VoIP introduces challenges in maintaining real-time voice quality, primarily due to jitter—variations in packet arrival timing—and subsequent packet loss. This study empirically investigates the effectiveness of Jitter Buffer (JB) algorithms in mitigating these issues across diverse commercial network environments. Using a rigorous pre-test/post-test design, we evaluated VoIP call quality by deactivating and then reactivating JB mechanisms on the FreePBX platform. Utilizing standardized metrics, including Mean Absolute Deviation (MAD) for jitter quantification and packet loss rate, and employing the G.711 codec with Linphone clients across three distinct commercial ISPs, we captured and analyzed performance under controlled conditions. Baseline measurements revealed significant jitter and packet loss variations across ISPs, with one ISP exhibiting particularly precarious transmission stability. Post-implementation, our findings demonstrate substantial improvements across all ISPs. Notably, packet loss was reduced to 0% across all packets, and MAD values decreased dramatically, indicating significantly enhanced temporal stability. These results, further translated through the Mean Opinion Score (MOS) framework, confirm JB's critical role in stabilizing VoIP transmissions, validating its efficacy as a generalizable solution for improving voice quality and user experience in practical, real-world network conditions, irrespective of underlying ISP infrastructure disparities.]]>
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