Twilio has introduced a new parameter called `jitterBufferSize` to its Conference feature, which allows developers to control the behavior of the jitter buffer and optimize the experience for participants with different network conditions. The jitter buffer is used to smooth out irregularity in media packet arrival times when mixing audio for conference participants, but it can introduce a fixed delay if not set correctly. With this new parameter, developers can now tune the behavior of their conferences to cater to different types of call quality issues and deliver the best experience for their users. The `jitterBufferSize` value can be set to small, medium, large, or off, with each setting having different effects on latency and audio artifacts. Twilio's tests indicate that changing the size of the jitter buffer reduces latency at the expense of increased audio artifacts, resulting in robotic and choppy audio being introduced instead of delay. Developers are recommended to use this parameter based on their specific requirements and types of calls, such as client or SIP participants, moderate or high jitter levels, and in-mixer vs transport latency considerations.