Month: February 2017

Balancing bandwidth considerations

When looking at a streaming WiFi audio system, there are several potential bandwidth bottlenecks. The Ethernet Switch The Venue Server The Wi-Fi Access Points IP Address space In our last technical blog, here, we looked at the Wi-Fi access points. Access points must be added to reach the total number of clients (smart phones and tablets) that will be listening at one time. Doing some quick calculations on data rates and bandwidth, assume 150 kbps for the communication to each phone and a 50% load capacity on the Ethernet, then 100 baseT (100 MHz) Ethernet can support about 300 phones and Gigabit Ethernet can support about 3000 phones, assuming no other significant traffic. It would presumably be best to use a cut-through switch for minimum latency, but as a practical matter, no one does this because of the cost. ExXothermic has designed a range of systems that can handle different number of clients depending on the processor and network interface, starting with 150 client phones for the most basic offering. Finally, one does need to get enough IP address space in the network so that each phone can grab one from DHCP. The usual netmask of 255.255.255.0, for instance, only provides for about 250 addresses. And don’t forget about the impact of DHCP lease times (we recommend 120 minutes). An IP address for a phone that has left the building...

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Load testing WiFi access points

We have recently been load testing WiFi access points for use with WiFi audio streaming. We had a lot of questions on the results because the specifications of enterprise grade WiFi do not really address our special case. One one hand, they talk about bandwidth, but we have very small packets and are not really bandwidth limited. For instance, an 802.11ac access point with an 80 MHz wide channel could theoretically support 3000 phone. That is way too high. And to maximize the quality of the audio, we set the bandwidth to 20 MHz. This minimizes interference with other WiFi devices as well as with other radios in the band. We do this because if one has high levels of interference, one hears the lost packets. Unlike a system like Sonos, we tune our system for minimum latency and best lip sync, which means we have no time to retransmit lost packets. In our system, one does not hear a single lost packet, but if there are several in a row, one does hear that. On the other hand, if one looks at the specifications for the number of VoIP (Voice over IP) calls an access point can support, one gets a very small number, e.g., 30 or 40. That is too low. VoIP traffic is not synchronized and people can talk at any time. In our system, there...

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