What we would need in the echolink system ist hat you can announce yourself
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to the system with a dtmf number ...
Example you are not at home but mobile in the range of any echolink node
Give that node your node number (with a command before) that the system
knows that you are not at home (reachable on your pc with your nodenumber)
but that you are "in the system" known to be reachable via repeater xyz
Now if any other wants to connect you then the system could answer with
"your xall will be forwarded to repeater xyz" ...
All echolink hams have a unique node number (that is your home pc) ...
But all hams also can be called with their callsign number (there is a code
for converting calls to dtmf numbers in echolink)
So ... all the routing and callsign forwarding is there already in the
echolink soft .... now what only is missing is that self-announce to the
system while mobile and not connected with your home pc
Ok ... no automatic routing like in a gsm net ... but something like
"my-bbs" in the packet net ....
Then integrating of some echolink nodes in the d-star (and maybe other) nets
would be a lot easier
Ps we made some tests with short timings (set to min) and with all spoken
things off (and no beeps etc.) with echolink .... nobody that did not know
it would hear any difference in the audio quality!!
[mailto:UniversalDigitalRadio@...] Im Auftrag von Matthew
Gesendet: Donnerstag, 24. Mai 2012 01:24
Betreff: Re: [UniversalDigitalRadio] Cross linking (Was: Codec2)
I think as long as the proper display is there, someone using the 1.2k
gmsk modem with analog audio (or echolink with callsign pass-through)
won't really be noticed. It will probably require some special tricks
somewhere along the line to properly work with the Icom gateway software
and such, but that's already taken care of in Jomathan's software.
On Wed, May 23, 2012 7:02 PM EDT John D. Hays wrote:
The argument that analog should never be linked to digital because ofwhite
noise and other artifacts forgets that Vocoders (as opposed to Codecs)like
AMBE and Codec-2 are specifically looking for speech and work hard tofor
remove background noise. All audio starts and ends as analog and
transitioning from an analog FM to digital voice system using a vocoder
applies the same filtering to the audio. I think the folks the argument
haven't tested their theory. Some who have tested it, can probably show
some issues like drop out, overdrive on CTCSS, etc. that will degrade the
system, however, well conditioned audio from an analog FM source should
encode much like analog voice from a microphone.
The second argument is often that the callsign information isn't there in
an analog signal or incompatible digital voice signal. Rules may vary
around the world, but here in the US the ID is for the transmitter where
the signal originates. It doesn't need to be passed through the network
identification purposes (other than courtesy or vanity) when it comes towhich
the regulations. In D-STAR, the callsign is also used as an address for
directed transmissions. In a linked system the only address needed is the
one that identifies the repeater(s) and conference channels, so that the
traffic is properly routed.
Any linking protocol that mixes systems should contain an ability to
identify the type of traffic source (Analog, Digital) and encoding (AMBE,
Codec-2, etc.) and allow the administrator of a repeater to determine
traffic to accept.
Given these two "non-issues", the rest is just politics. :)
John D. Hays
PO Box 1223, Edmonds, WA 98020-1223