Icecast - Jun 2007 - RTP/RTSP streaming of GSM or ADPCM audio

Greetings:

It would be nice if Icecast supported RTSP; however I would
appreciate any suggestions for a small RTSP/RTP solution to
encode 8kHz mono audio in GSM or ADPCM and service multiple
unicast client connections.  The ideal would be a black-box
hardware solution with an audio input and ethernet interface
similar to broadcast studio IP audio links or the network
audio capabilities of certain X-terminals.  An acceptable
solution would be *nix or Win32 software using PC hardware.
The closest I've found on the 'Net is the old Ulm WebMedia
software from 1997 (win32) but it seems to use a non-standard
RTSP implementation that requires a custom player. There
is a lot of talk on Java discussion forums about coding
this sort of app, but I'd prefer something that did not
need the high overhead of a JVM.

One could use the CCRTP libraries to build such a program
and sample code exists that creates a sender and listener
but nothing like a stream server seems to exist.

For the moment, I've kludged up a SIP UA to service a few
incoming calls in a conference, but that won't scale to a
practical number of listeners.  The Columbia rtspd would work
but does not support GSM or ADPCM without modification; I
don't believe that Apple's server does either.

All replies are much appreciated.

Regards,

Michael

Michael Grigoni wrote:
> Greetings:
>
> It would be nice if Icecast supported RTSP; 
It probably never will
> however I would
> appreciate any suggestions for a small RTSP/RTP solution to
> encode 8kHz mono audio in GSM or ADPCM and service multiple
> unicast client connections.  
why not use icecast (with adjusted buffers) + speex? are you really that
reliant on very low latency?

Cheers

Thomas

Thomas B. Ruecker wrote:
> Michael Grigoni wrote:
> 
>>Greetings:
>>
>>It would be nice if Icecast supported RTSP; 
> 
> It probably never will
> 
>>however I would
>>appreciate any suggestions for a small RTSP/RTP solution to
>>encode 8kHz mono audio in GSM or ADPCM and service multiple
>>unicast client connections.  
> 
> why not use icecast (with adjusted buffers) + speex? are you really that
> reliant on very low latency?
> 
Thanks for your reply.  Yes, the application is an online remote-
controlled HF receiver and a DX'er can't tolerate lag in the audio
while tuning.

I'd appreciate hearing from folks who have solved this problem.

Regards,

Michael