Technical Highlights
Best interface available for all wired and wireless telecommunication systems
Wide range of coding algorithms, including AAC and OPUS
IP interface for SIP/AoIP, compliant with EBU Tech 3326 (N/ACIP)
Integrated 2G/3G/3G+/LTE module (optional), supporting HD-Voice (AMR-WB)
"Direct RTP" mode
3 mic/line inputs, 2 stereo headphones, 2 line outputs
Highly integrated unit, lightweight and compact
Intuitive, user-friendly interface
High performance audio interfaces
2 antennas in diversity
Recording capability
DC or batteries, autonomy > 5 hours
Inmarsat BGAN integrated application
USB for audio interface with a computer
Coordination features, using the mobile network
Scoopy+ S right side viewScoopy+ S left side viewScoopy+ S rear view
Operational benefits
With Scoopy+ S, you never lose a story, as this lightweight portable codec not only transmits from everywher to everywher, it also offers recording and editing functions. Over IP, ISDN, POTS or the integrated mobile network module for HD Voice, GSM, 3G and 4G/LTE, you are always able to perform your live transmission.
The Scoopy+ S is a highly versatile audio codec and can be used in a carrying bag for mobile news gathering or on a desk for sports and other events.
In IP mode, the unique packet duplication with interleaving ensures a highly robust link. Recorded audio files can be transmitted via FTP at high speed over 4G/LTE networks or other IP links.
The Scoopy+ S is N/ACIP compatible and can be configured easily through its internal website. A large selecion of algorithms including G722, MPEG Layer II, AAC, OPUS and cellular phone calls in HD Voice quality are available.
With its "HD-4G" option that provides an integrated wireless module and two antennas, Scoopy+ S natively supports both voice service (and HD Voice) and IP transmission over 2G/3G/3G+/LTE networks. By connecting a USB mobile key it can also connect to a second mobile network. See our FAQ and check the supported 3G/3G+ and LTE devices.
"Direct RTP" mode allows you to setup an audio call over an IP network without using the Session Initiation Protocol (SIP). Please download this application note to see how to operate the codecs using the "Direct RTP" mode.
Ability to setup an audio call over an IP network in “Multicast” mode. The “Multicast” feature is available in the “Setup / Network / AoIP Parameters” menu fro the front panel or in the “AoIP” page of the web interface.
Frequency bands supported
To know the bands used by the operators in a given country, you can for instance check on this site.
You can find a list of the network operators that support the "HD voice" service in the section "Information Papers" besides.
