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The AETA AUDIO SYSTEMS codecs currently on the market, as far as they are fit with the "Wireless" option for accessing mobile networks, all support the HD Voice service.
Mobile operators usually sell "HD" calls at same rates and conditions as standard telephony, and the service is automatically included in voice subscriptions.
As this is still circuit switched telephony, calls are charged based on duration (or fixed price, etc.) just like normal voice service.
Provided that the network and the involved terminals support AMR-WB, the terminals will encode/decode using this algorithm.
The switch to "HD Voice" is automatic; no special handling to do, no special subscription needed. On the contrary, it is not normally possible to prevent this from taking place, and this is not signalled neither! Only the user's ears will tell...
"Folding back" to narrow band takes place when one of the terminals is not compatible, and/or there is an interconnection with a non-compatible network (a noticeable example is the "wired" telephone network). So far there is no connectivity between competing operators or different countries (as of june 2011). This will eventually be implemented, however.
AMR-WB (=Advanced Multi-Rate Wide Band) is a wideband (50-7000 Hz) speech coding algorithm, to be compared with the common narrow band speech codecs (300-3400 Hz). AMR-WB is also standardised as G722.2 by the ITU-T.
AMR-WB is used by the service called "HD Voice" by mobile operators.
HD Voice is not a VoIP service, although this is suggested by some features. This is a circuit switched mode, as is the basis for the mobile telephony service often called "GSM". For deployment reasons, most operators implement this new service on their 3G/3G+ base stations, and less often on the 2G (GSM) ones. As the sales communication emphasises the Internet and data features for the 3G/3G+ networks, users may forget that these networks also support the circuit mode telephony service, just like the 2G networks.
Transmission of HD Voice uses same channels and benefits same conditions and quality or service as "standard" mobile voice. The priority granted to voice conversations (over packet mode data) in high traffic situations is still valid, and the network resources allocated to a link stay committed until it is released, even if other users are soliciting the network.
The bits rates used are roughly similar to those used with GSM, the bandwidth extension is due more to the codec performance and efficiency than a higher bit rate.
As the primary goal is telephony, the system and coding are designed for a moderate latency, close to that of normal mobile telephone communications.
Plugging a USB device is one way you can add mobile access to a Scoopy+ or a Scoop 5. This can be useful in several ways:
* Add mobile IP to a unit without wireless capability (consult us to get the software upgrade)
* Add an alternative network when the unit has got the wireless feature
* Get LTE connectivity on a wireless unit
The device must be supported by the codec. Currently supported USB devices:
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This is a recommendation from the EBU, aiming at interoperating audio codecs over IP networks. The EBU recommends the SIP and SDP protocols, as well as a mimimum set of coding algorithms to be supported. Our Audio over IP codecs all comply with this recommendation, and they interoperate with many other brands.
Using a server is not mandatory with the SIP protocol, and you can definitely set up a point-to-point link without such server. Just use the IP address of the destination codec to set up the link. There is no need to disable the SIP protocol! Even not using a SIP server, you still benefit from the automatic negotiation of the coding parameters, and do not need to handle details such as UDP port numbers, etc.