Dropouts

@Oele.
I slightly modified Firefox's User-Agent by adding the string "iPhone" to make it <strong>Mozilla/5.0 (X11; Linux i686; rv:2.0.1; iPhone) Firefox/3.6.9</strong> and M.YOURMUZE then supplied me a file download <strong>p.aac</strong> which I could play with mplayer. The download speed was at around 5-6KB/sec which looks like 48kbit/sec so it came in about twice as fast as the player will play it. I calculated mplayer's playing speed used up around 21kbit/sec which is near what I'd set for my YOURMUZE bit rate. So this looks promising. Sure are no dropouts when I'm playing from a file! it is important that the letter P in iPhone be capitalized! :-) I settled on slightly modifying the string as above so that I can readily set it back to the right Firefox string (to satisfy the rest of my browsing tabs) as soon as YOURMUZE streaming commences--and I do that by merely deleting the i in iPhone. I found that leaving the string as Phone is inconsequential; it's not seen as anything to do with iPhone.

When I play any stream in my list I get small dropouts through the whole stream. I tried changing the bit rate to other settings to fix this problem, but it doesn't do anything.

@ ofelia526:
The answer to your problem lies in this thread. When you are listening on a mobile device then it is inevitable that wireless internet will lose some packets of data. But if you are using a device with a proper browser, you can tell that browser to pretend it's an iPhone (using the method in my post just preceding yours in this thread) and YOURMUZE will then fill in the missing packets so your program plays with no discernible dropouts. Otherwise, if you don't have an iPhone, or a proper browser, then you are out of luck. Sorry, nothing can be done to improve it apart from trying a better 3G phone service provider with less congestion.

Oele, your neat trick seems to no longer work. If I set firefox's user agent to masquerade as an iPhone I still get the rtmp stream. I REALLY REALLY would like the .aac stream for my PC to avoid rtmp's aggravating gaps of silence.

Is there some other trick I can adopt to avoid rtmp? Why don't you make it easier for users to select the tcp stream? Perhaps by a setting on our login page at m.yourmuze.fm?