Telos Zephyr Xport - Manual (Page 41)

ZEPHYR XPORT
USER’S GUIDE
CHAPTER 4 - FREQUENTLY ASKED QUESTIONS
31
Is there anything I need to know about using ISDN for this application?
Nothing special. The telephone network automatically translates the signals on the analog line to
the digital form (called "G.711") carried over ISDN.
The ISDN line must be configured to allow this type of call (called Circuit Switched Voice or
"CSD"). However, the vast majority of ISDN lines have this feature (our ISDN order forms have
always requested this feature).
Can I share an ISDN line between two Xports?
Yes. As with many ISDN terminals, you can connect multiple Xports together using the S interface.
See section 3.6 for more information.
How does the studio Zephyr know what kind of call is being received?
First, we look at the ISDN “set- up message,” that tells us if the incoming call is from a digital
device or from an analog phone line. Then we wait for the modem tone. Then we initiate Xport-
specific handshaking to be sure.
Will it work for both local and long-distance calls? Anything special I need to do?
Within the USA and most other countries, all switching and transmission of telephone calls are
done digitally, so performance should be nearly as good for long distance as for local. In both
cases, only the “last mile” analog part should have any effect on modem performance.
What about international calls?
You should experience better quality and fewer drop- outs with the Xport approach compared to
POTS- only codecs. Most international calls are switched and transmitted over digital links, so the
only analog part will be the last mile at the remote site.
On some connections, however, there may be some compression used to conserve bandwidth.
The most common is ADPCM, which reduces the usual Telco 64 kbps rate to 32 kbps. There may
also be “echo- suppressors” at some point within the connection. If either of these occurs, you
may want to try a different long- distance carrier to get a good connection. Note that these
problems are possible with POTS- only connections also. Actually, ISDN makes them less probable
because the Telco network is likely to assign better facilities to digital calls.
What happens when the line is very bad?
In this case, modem bitrate will be too low to be useful, and the Xport will switch to a non- modem
“audio coupler” mode. Because we use digital hybrid technology borrowed from our telephone
interface products, send/receive isolation is quite good. You will not get high- fidelity audio, but
you will get the best possible. See Section 5 (What to do if…) for troubleshooting tips.
What is the reasoning behind your choice of the aacPlus audio coding method?
It is, without much doubt, the best low- bitrate codec there is. MPEG AAC has
been independently tested using a double- blind procedure and found to be
superior to any other scheme at rates down to 16 kbps. aacPlus uses a
powerful spectral band replication method to take this already excellent
performance to jaw- dropping amazing. Because it is an enhancement
designed specifically for very low bitrates, it is perfect for POTS codecs. (As a side note, SBR
added to MP3 is called MP3 Pro, and is catching on quickly for Internet applications.)
aacPlus in being considered for inclusion within MPEG, and will probably be adopted within the
next few months. XM Satellite Radio has announced that they are using it.
Other POTS codecs use proprietary coding methods that have not been independently tested, and
are likely to be much worse than aacPlus. Some use CELP, which is a voice- only codec – basically a
scaled- up version of the codec in mobile phones. These don’t work very well for music or for voice