Streaming does not seem to work reliably. Sometimes, audio does not arrive at all or only with delay.
Check if you have sufficient bandwidth for streaming. Streaming uncompressed data at a constant bitrate requires a certain bandwidth which you can calculate. Having other network traffic going on at the same time reduces this bandwidth. In WLANs, other network nodes may also reduce the bandwidth. You can conserve bandwidth by reducing the audio format quality, e.g. by choosing a sample rate of 16 KHz instead of 48, or using mono instead of stereo.
Repeated data loss can also cause delays and drops. If you are using a wireless connection with bad transmission conditions (distance, obstacles, interference…), you are likely to experience drops. This is particularly bad with TCP-based streaming (including HTTP streaming), since each drop will cause streaming to block until a retransmission occurs.
Make sure that your client device has enough processing power. We have observed that interruptions in streaming mistakenly attributed to the network can also be caused by an endpoint device with low processing power (Google Glass), which was unable to handle the streaming at high bitrate in parallel with other tasks.
Generally, the streaming monitors (configuration pages for streaming virtual devices / servers) in Audio Manager will help you analyze throughput and identify the bottleneck of your streaming.