Firmware Scoopy+S

Improvements :

• Improved the AoIP interoperability with third-party units using the “MPA” payload type.

Fixed issues:

• When establishing an AoIP call to a Cellphone, if the callee picks up the incoming call after more than a certain delay (approximately 15 seconds), it results in a “Waiting for Sync” for the call.

Fixed issues:

• After a software upgrade of our STUN Server, the NAT detection process is sometimes returning an erroneous NAT type.

Improvements :

• From the “Network > AoIP Parameters” page of the html interface, ability to configure the method to process the SIP keepalive messages, this may reduce the overall SIP communication with some SIP servers.

Improvements :

• In the “AoIP” page of the html interface, refresh the “Secondary SIP Account” group-box when the “Alternate SIP Port 2” setting is modified.
• The APN for “Orange FR” mobile network is now defaulted to “orange”.

Fixed issues :

• In Wi-Fi mode, fixed several bugs.

New features :

• Added support for the AAC-LD and AAC-ELD coding algorithms (AoIP).

Improvements :

• Possibility to send control codes as prefixes of the dial string for AoIP and ISDN calls. The * and # characters usually delimit the control code, the network or the PBX uses this control code to trigger certain actions.

Improvements :

• Improved locating SIP servers when associated with a single domain name (as described in RFC 3263), i.e. send a new request to the next SIP server in the list when a SIP registration failure occurs.
• Ability to configure the “session timers” mode (as described in RFC 4028) from the “Setup -> Network > AoIP Parameters” menu of the front panel and from the “Network > AoIP Parameters” html page; enabling the “session timers” may improve the interoperability with some SIP servers, this mode is disabled by default.

Fixed issues :

• Regression introduced in version 1.17: when updating an existing call profiles from the front panel or the html interface, it is saved as “unnamed”.

Fixed issues :

• Booting the codec will mistakenly restore the default APN of the Mobile module.

New features :

• Added support for the “Auto Bit Rate” feature for the Opus coding algorithm.
• Added an adaptive mode to the jitter buffer for AoIP calls.

See the application note “How to use Auto Bit rate Opus and Auto Jitter Buffer“.

New features :

• Up to two internal mobile modules supported with the new “dual mobile” version.

Improvements :

• Support of VoLTE (mobile voice over 4G) with “HD Voice” interoperability between networks.
• Improved interoperability with more SIP servers, with added support for “session timers” (as described in RFC 4028).

Fixed issues :

• A networking issue occurring during the setup of an AoIP call could cause the codec to remain blocked in a “connecting” state until the call was manually released.
• Restored the recording function, defective on release 1.15.

New features :

• As an alternative to “Double streaming” which secures an AoIP/SIP link with two redundant paths, support of a “Bonding” scheme that distributes the AoIP stream over two lower bit rate channels. More information in this application note.

Improvements :

• The AoIP quality statistics now remain displayed on the “Connections” html page even after the termination of a link.
• These statistics are available separately for each of the streams involved in a link using “Double streaming” or “Bonding”.

Improvements :

• Ability to perform double streaming AoIP calls using two SIP servers. This feature is rarely useful, but it allows easily switching between single codec and double streaming modes.
• Warning message when attempting to change the “Preferred Network Operator” setting while the data service is active for this mobile controller.

Fixed issues:

• Erroneous tooltip information on the “Remote Access Portal” when a device was registered simultaneously on two SIP servers.

New features :

• Ability to register simultaneously on two SIP servers so that the unit can accept incoming AoIP calls indifferently from either server. When using this feature, one of the servers is selected as default to make outgoing calls.
• Ability to filter incoming AoIP calls and accept only incoming calls originated from the configured SIP server(s).
• New “Advanced Settings” in the “Coding” page of the web interface: ability to hold an AoIP audio connection even without receiving packets, ability to configure a 6 dB attenuation for the G711 decoder.

Improvements :

• New “Redial/Release” function to control the state of an audio connection from a single GPI.
• Ability to reset the call history in the “Reset” page of the html interface.

Improvements:

• Improved communication with the remote access server for a faster information update on the “Remote Access Portal”.

Fixed issues:

• In some LAN configurations, the IP routing could be defective and lead for instance to a SIP registration failure.

Improvements:

• Increased reliability for the firmware update process.

New features :

• Ability to manage SIP profiles, from the “Directory” menu of the front panel or the “SIP Profiles” page of the html interface.
• Ability to manage the equipment using SNMP (Simple Network Management Protocol); note that only a subset of the features and information is exposed to the SNMP manager.

Improvements :

• Ability to use an “alternate” SIP Port for the secondary SIP account used in double codec mode or for double streaming.
• Added support for new external LTE/4G USB devices: Huawei E3372h-153 and Huawei MS2372h-153.
• Support of MPEG Layer III (AoIP).
• “Transmission quality meters” displayed on the frontpanel for a Wi-Fi connection.

Fixed issues :

• Device blocked if the USB Wi-Fi adapter is removed while Wi-Fi is set as default network.
• Audio routing issue while in cell phone mode if an audio input setting is changed via the html interface.

Fixed issues:

• Embedded web interface: in case the network for remote access is supported by a removable device (e.g. 4G stick) and it becomes unavailable (device unplugged), an erroneous “Auto” was displayed.

Fixed issues:

• Loss of the default IP route of the Ethernet interface whenever loading a Preset that involves static IP addressing for this interface, causing a SIP registration failure.

New features :

• Added support for the AETA “Double Streaming” system: ability to secure an audio over IP link by establishing two AoIP connections containing the same audio stream via two separate routes.
• Added the ability to connect on a Wi-Fi network, using a USB adapter device.
• Ability to import/export profiles from/to an SD card.
• Support for an AES audio source on Input 3 (new hardware option).

Improvements :

• Added support for AAC-LC and HE-AAC dual mono coding at 48 kHz (AoIP calls).

Fixed issues :

• Web interface: in the “Audio” page, the “Gain” adjustment was automatically disabled if the “PAD” was set.

Fixed issues :

• Regression introduced in version 1.10 : failure of the domain name resolution in some network access configurations (public DNS server address). As a consequence, malfunction when a domain name is used for the SIP or the STUN server.

Improvements :

• “Remote Assistance/Access” service: ability to select the network interface that will be used for remote access (“Tools/Maintenance/Remote Access” sub-menu of the front panel, or “Maintenance/Remote Access” html page).
  The “Auto” choice is still available, in which the interface is always the same as the one used for audio transmission.

Fixed issues :

• After a power cycle, failure to “redial” by using a contact input (GPI).

New features :

• Added support for the “Remote Assistance/Access” service. The settings and state of this service are accessible from the “Tools” menu or the embedded html pages.

Important notice: if the current firmware is older than 1.08 and a password is set for accessing the html pages, you MUST reset it to a blank value (refer to page Maintenance/Login data), BEFORE performing this firmware update.
Once the update is completed, you can later set a password again.

Web pages security improvements :

• Implementation of HTTPS for the embedded server.
• Enhanced protection against attacks.
• Implemented a technique to prevent from Cross-Site Request Forgery attacks.

Improvements :

• When importing a configuration file, the login password is not affected (previously, the password possibly included in the configuration file was also imported).
• Opus coding is upgraded to version 1.2.1.
• Support for following 4G/LTE USB sticks: Huawei E392, E398 and E3276, Vodafone K5005.

Improvements:

• Improved handling of SIP call redirection.

Fixed issues :

• If an incoming call is received while an AoIP session is running, call rejection with a “603 Decline” SIP code instead of “486 Busy Here”. Issue on version 1.07.01, fixed.

New features :

• Ability to select distinctive network quality settings for the Ethernet and the mobile IP interfaces.
• Availability of LOAS encapsulation for AAC coding algorithms (for ISDN transmission), upgrading the interoperability with third-party codecs.

Improvements :

• AoIP transmission statistics displayed on the “Connections” page of the html interface.
• Ability to dial a number in the restricted access mode. From the default screen (bargraphs) one can dial and call a number. From the “Main” menu a pop-up message asks for the password granting full access to all the menus.
• The Scoopy+ S now reboots right after a “Factory Reset” is performed.
• AAC algorithms included in the codecs detected by the 5AS system.
• More explicit error messages when an outgoing call fails.

Fixed issues :

• Inaccurate overload indication on html page.

Fixed issues :

• When recalling an ISDN call profile, the current coding algorithm was kept instead of recalling the one included in the profile.
• ISDN line disturbed if another codec is connected on the same ISDN bus.
• While recording, the elapsed time and remaining time counter would increase/decrease by steps of two seconds:  the increment is now one second.
• Web interface, “Call Profiles” page: when clicking a profile in the list, the “Network” parameter was forced to “Ethernet”. Fixed, the network recorded in the profile is recalled instead.

New features :

• Ability to customize the title of the tab in the html browser; this allows to identify easily a unit among a number of open pages on the browser.
  This new settings can be found in the “Misc” html page.
• For mobile data, it is now possible to select between the “PAP” and “CHAP” authentication modes.

Improvements :

• Improved the AoIP interoperability with third-party units that deal improperly with the channel number in SDP.
• Improved AoIP interoperability for Opus coding, especially regarding the management of the stereo format and the bit rate.

Fixed issues :

• Interoperability issue with codecs using the “MPA” payload type: fixed.
• The device hangs if an ISDN link is set with AAC but with dissimilar configurations (different type AAC-LC or HE-AAC, or mono vs stereo): fixed.
• Fixed various malfunctions in the handling of presets (create/update/load), in some cases imposing to reboot the unit.
• Default route deleted from the routing table after editing the Ethernet parameters, leading to loss of control from a remote network: fixed.

New features :

• Ability to setup an audio connection 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 connection over an IP network in Multicast mode.

Improvements :

• Improved the stability of the AoIP controller for both Ethernet and wireless IP interfaces.
• Display of the transmission quality on the root menu during IP communication for all algorithms (not only OPUS): 6-segment indicators showing respectively the quality of the transmission and the reception on the network.
• Ability to display the “Relay status” in idle mode from the root menu when pressing the joystick to the right.
• Web interface: added a title in the help section to remind the active tab. Front panel: show the “SIP User” parameter in the “About” menu.
• Addition of two new audio routing modes for mono operation: “Standard” (default) or “Main & Backup”.
• GPIO management: ability to manage two GPIO. This feature is available from the front panel (Tools/Misc/GPIO) or on the “Misc” page of the web interface.
• GPIO extension: ability to manage four additional GPIO by using an extension USB relay box. This feature is available from the front panel (Tools/Misc/GPIO Extension) or on the “Misc” page of the web interface.

Fixed issues :

• With very bad transmission quality, changing the bitrate of the Opus encoder on the fly, or using the “Relay transmission” feature, could release the call.
• Under certain circumstances, an attempt to setup an audio over IP call with the packet replication option activated could lead to a failure to synchronize the audio.
• Random sync losses with G722 SRT via ISDN lines emulated over an IP network.

New features :

• Type of NAT displayed when STUN is active (“About” menu on front panel, “Status” page of html interface).
• Ability to change the Opus bitrate during transmission, either via the front panel interface or the embedded html pages.
• Transmission quality bargraph indicators displayed on the front panel
• Support for the Thrane & Thrane Explorer 510 terminal (BGAN option)

Improvements :

• For mobile IP data, improved handling of data connection issues (failure to establish a connection, unexpected data disconnection, connection retry process).
• Increased resilience to IP packet losses for G711, G722 and MPEG L2.
• “Remote” and “Local” profiles renamed, become “Call profiles” and “Presets”.
• Improved stability of the starting process when there is no IP network connectivity
• Screen saver period reduced to 5 s.
• On html page, parameters related to mobile data (data activation, APN, network quality..) are now accessible even if the current mobile mode is “Voice”.

Fixed issues :

• Improper decoding of some SMS messages.
• BGAN terminal detected but communication failure; communication over BGAN now fully operative.
• Selection of IP or voice mode for mobile network in call profiles.

First released

Not existing