NOTES, RESOURCES, ADDITIONAL INFORMATION | 93
(VoIP) does not distinguish between “VoIP over the Internet” versus “Voice over other (man-
aged) IP networks”. If you have been keeping up with the transition to IP Codecs, you probably
have noticed the same terminology issue there. IP-based STLs over an IP T1 are just as reliable
as traditional STLs over traditional TDM T1 circuits. You are completely right to be concerned
about a VoIP trunk (or an STL) over the Internet, as that is not at all the same thing, and perfor-
mance in that case could be variable. Inside the facility, on a LAN, all problems disappear, since
you have plenty of bandwidth and full control over the network.
So shared bandwidth is the only problem with VoIP?
Well, there’s audio quality. In the early days VoIP used a lot of compression, with bit rates
being as low as 6kbps. Needless to say, the resulting audio was not impressive. These low-rate
codecs have mostly fallen by the wayside. The lowest-grade codec the VX supports is g.711,
the standard for digital audio in the telephone network pre-IP. And you will eventually benefit
from higher-fidelity codecs as these proliferate in the VoIP world.
How does AoIP relate to VoIP?
Despite the similar names and underlying technologies, they are very different with regard to
performance and application. An analog phone line and a balanced 600Ω studio audio circuit
are pretty much the same tech, but the applications and performance are very different. ’AoIP’
has come to mean professional studio-grade audio networking - full-fidelity and usually with
no compression. Low-delay and synchronized channels are other distinguishing characteris-
tics. Another way the two differ is that AoIP uses an advertising/discovery protocol for receiv-
ers to find sources instead of the Session Initiation Protocol (SIP) that VoIP employs. AoIP
uses a system-wide clock mechanism to support low-delay and tightly synchronized channels.
Finally, AoIP often takes advantage of IP’s multicast capability to permit multiple receivers to
listen to an audio source efficiently. AoIP is intended for managed, guaranteed-bandwidth
networks, such as with an Ethernet switch as the core of a local area network.
And what about ‘IP codecs’?
Now you are very close to VoIP! These use SIP for call setup and various codecs for compres-
sion, so are similar to VoIP telephones. In fact, they actually are VoIP telephones and can
sometimes interoperate with them. They have better codecs than VoIP phones, though. AAC-
ELD, in particular. Advanced IP codecs, such as our Z/IP, employ sophisticated technologies to
overcome the Internet’s deficiencies. Dynamic buffering, error concealment and more clever
stuff.
I have read that some VoIP PBXs use IAX trunking. Can I use the VX with these?
IAX is a protocol invented by the Asterisk people. It provides functions similar to SIP, but with
more bandwidth efficiency. The VX doesn’t support IAX trunking at this time. But you can
connect the VX to an Asterisk with SIP trunking or as multiple SIP extensions. There’s plenty
of bandwidth on a LAN, so this works fine, while staying with a standards-based approach. We
like Asterisk as a VX adjunct. It can add voice mail, automated attendant, blocking callers from
caller ID, off-premise SIP extensions, and more, to a VX installation. Asterisk is free Linux-
based PBX software that runs on a PC. The VX and Asterisk PBX are an attractive combo we
expect will become popular within the broadcast industry.
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