I use Wowza RTSP server to read AAC audio file. The file is packitized as ts packet then sent via udp.
The problem i s that somtimes my audio client(which recieves , decodes and displays stream) triggers a timeout . The timeout means that it doesen't receives enough data to fill a buffer then start decoding during 15s.
Analyzing wireshark capture i have found that when this timeout is triggered the bitrate is less then 0.019MB/S wheras it is equal to 0.33MB/S when stream is launched normally.
I use a private network for test and i can't modify the audio client.
Has any one see this problem? can i increase server bitrate ? is the lowbitrate caused by transcoding stream?