NOTES, RESOURCES, ADDITIONAL INFORMATION | 51
In a studio equipped with an Axia console, the needed signal connections are available via
Livewire. Configure both the console and the VX Engine with LW channel assignments so that
the correct connections are made. In a non-Axia studio, route the AEC connections to a LW Node
by assigning the appropriate LW channels at both ends, and then connect the audio to/from the
mixing console via analog or AES3 to the Node.
To check if an AEC is working, call into the system and switch on the mic and loudspeaker.
Adjust the mic level to normal and the loudspeaker to reasonable volume. Talk into the phone;
listen for echo and assure that none is audible.
Telco Services and Interfaces
VoIP: SIP Trunking
While it remains a niche in early 2011, SIP trunking is growing rapidly in support from both
PBX vendors and carriers. Over time, this will almost certainly appreciably reduce the use of
the older POTS and T1 trunking. Eventually, it may replace it completely.
Whether the gateway to the PSTN is at your physical location or at another site should make
no difference – as long as the IP path between you and the gateway has guaranteed QoS with
sufficient bandwidth to support the maximum number of active connections you expect to
have. In the case that the IP link is to be used for both telephony and data, the system must be
designed so that phone calls have priority. In order to ensure this, there must be only one IP
vendor between you and the PSTN, and this vendor must guarantee QoS in a properly written
Service Level Agreement. Any time that IP service crosses from one vendor to another, all bets
are off as to both the probability of achieving consistent good quality and having any chance of
getting problems resolved.
The other thing to look out for is what codec will be used. For calls that ultimately are carried
by the PSTN, only the native g.711 codec is acceptable for broadcast applications. Anything
else would involve transcoding and an unacceptable reduction in fidelity, especially audible
when mobile phone calls are involved. These already have poor quality due to their low-rate
14.4kbps codec. Passing this through g.711 within the PSTN and then yet another codec on the
way to your studio over an IP link is asking for aural trouble.
The VX natively supports the g.711 A-Law, g.711 µ-Law, and g.722 codecs. Almost all PBXs and
gateways support these formats.
Finally, you need to be sure that your equipment and the carrier’s gear can properly communi-
cate. While SIP is a standard, vendors often enhance it with extensions that are not universally
supported.
One development that could help is a project called SIPconnect, undertaken by SIP Forum, a
consortium of SIP vendors. The SIPconnect Interface Specification was launched by Cbeyond
Communications in 2004 with support from Avaya, BroadSoft, Centrepoint Technologies,
Cisco, and Mitel. It attempts to detail the interconnection specifications between IP PBXs and
VoIP service provider networks. It specifies a reference architecture, required protocols and
features, and implementation rules. It calls for the g.711 codec to be provided on all equipment
and services.
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