Firmware µScoop

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.

Fixed issues :

• In double codec mode, with a first AoIP call established to a Cellphone with the knock-on feature enabled, the attempt to setup a second AoIP call to this same Cellphone, and if the callee decides to pick up the second incoming call, results in a “Waiting for Sync” for the second call. No subsequent call will succeed, requiring a reboot of the codec.

New features :

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

Improvements :

• Possibility to send control codes as prefixes of the dial string for AoIP calls. The * and # characters usually delimit the control code, the network 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 “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 1.13: when updating an existing call profiles from the html interface, it is saved as “unnamed”.
• On the “Digital” version of μScoop, improved AES synchronization when input signal has low audio level.

Fixed issues :

• Don’t hang up the active call when the unit is rebooted from its web interface, this will stop the audio redial mechanism.

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“.

Improvements :

• 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.

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.
• Support of the AARC protocol for remote controlling the µScoop.

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”.

Fixed issues:

• Regression introduced in version 1.10 : GPO2 does not follow the received “Info 2” if GPO1 is not assigned the “Info 1” state.

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.

Fixed issues:

• Erroneous tooltip information on the “Remote Access Portal” when the unit 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.

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 “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.

Fixed issues:

• The state of the relays (GPIO) was not automatically updated in the “Status” page of the web interface.

Fixed issues:

• Embedded web interface: in case the network for remote access is supported by a removable device (e.g. Ethernet USB adapter) 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.

Improvements :

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

Fixed issues :

• Regression introduced in version 1.06 : 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 (“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 for this service are accessible on the “Maintenance” page of the html interface, and its current state is visible on the “Status” page.

Important notice: if 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.

Improvements:

• Detailed statistics of the AoIP transmission displayed on the “Connections” page of the html interface.
• Improved handling of SIP call redirection.

Fixed issues :

• GPI not operative when configured for redialling or releasing the link: fixed

Fixed issues :

• Default route deleted from the routing table after editing the Ethernet parameters, leading to loss of control from a remote network: fixed.

Improvements :

• All the LEDs flash briefly during startup.
• The “Ready” and “Dec” LEDs blink red during the firmware update process (update from version 1.02).

First released

Not released