A key issue for successfully interconnecting musicians in real-time over the Internet is minimizing the end-to-end signal delay for transmission and coding. Anyhow, the variance of transmission delay (``jitter`) occasionally causes some packets arrive too late for playback. To avoid this problem previous approaches are working with rather large receive buffers while accepting larger delay. In this paper we will present a novel solution that keeps buffer sizes and delay minimal. On the network layer we are using a highly optimized audio framework called ``Soundjack` and on the coding layer we are working with an ultra low-delay codec for high-quality audio. We analyze and evaluate a modified transmission and coding scheme for the Fraunhofer Ultra-Low-Delay (ULD) audio coder, which is designed to be more resilient to lost and late arriving data packets.
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https://aes2.org/publications/elibrary-page/?id=14272
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