[Icecast] RTP/RTSP streaming of GSM or ADPCM audio

[Thomas B. Ruecker]

Michael Grigoni wrote:
> 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.
Then icecast is clearly the wrong option. I have used it to live
broadcast some QSOs I've made - the listeners of course were
non-interactive. DD6VSF also had a RX controlled through a CGI interface
that could be tuned in to via icecast. Absolutely sufficient for the
occasional enigma check or pileup cross-check.

Using something like callweaver or sipX should be a better option then.
The signalling can be used to control the RX and also give feedback to
the listener connected (current QRG, mode, filter). Conference servers
could be used to connect multiple listeners.

vy 73 es 55 de Thomas, dm8tbr