|
Optimize IP Connections |
Top Previous Next |
|
Problems at IP-audio-connections between
1. Increasing the IP audio delay on the decoder side The IP audio delay can be set from 0 ms (default) up to 5000 ms at menu item Settings/Network. Increasing the audio delay means that first all received audio data is buffered before it is decoded. Such buffering helps to prevent audio interruptions and distortions caused by jitters. The bigger the receive buffer the more reliable the audio IP transmissions is. On the other hand the audio delay increases.
2. Adapt transfer rate and duplex mode Sometimes interruptions or distortions of IP audio transmissions are caused by improper auto sensing of the network equipment (switch, hub , router etc.). In this case the transfer rate and the duplex mode should be changed ‘auto’ to the settings of the network. This can be at menu item Settings/Network. Example: With some network switches it is useful to change
3. Adjust Type of Service
4. Adapt packetsize Sometimes interruptions and distortions of IP transmissions are caused by the fact that the upload and/or download capacity is too small. Therefore it should be considered the really needed bit rate is higher than the bit rate of encoder due to the IP overhead. At UDP transmissions this IP overhead does not fluctuate and can be calculated exactly. More details about UDP/IP overhead calculation can be found in chapter IP Overhead. Usually the bigger packet size is the smaller the IP overhead is. However, there is one restriction: If the payload (audio data) plus the IP overhead is bigger than max. possible packet size value allowed by the Therefore the The IP packet size mode can be set at menu item Settings/Network to (provided system software version 2.2.0.0 or later is used ):
|