Firmware Scoop5 S-IP

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.

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 1.17: when updating an existing call profiles from the html interface, it is saved as “unnamed”.

Improvements :

• In double codec mode, improved management of incoming AoIP calls and routing to the appropriate codec when the callee is addressed with an alias instead of the main SIP URI.

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.

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, double codec and double streaming modes.

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 :

• 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 “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.
• Support of MPEG Layer III (AoIP)

Fixed issues:

• Embedded web interface: in case the network for remote access is supported by a removable device (Ethernet/USB adapter) and it becomes unavailable (device unplugged), an erroneous “Auto” was displayed.

Fixed issues:

• Loss of the default IP route of an 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 in single codec mode).

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.
• Defective operation of the auxiliary data channel (LL transmission).

Fixed issues :

• Switching to double AoIP codec mode used to fail on version 1.10 when using the html page “Network / AoIP parameters”.
  This issue is only relevant for units fitted with the “double AoIP codec” option.

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.
• In double codec mode, ability to use loop control on Codec 2 for releasing a running connection over Ethernet networks.

Fixed issues :

• The “Auto redial” feature fails to restore a link after a power cycle.
• The “Backup call” variation of the “Loop Control” feature was not operative.

New features :

• Added support for the “Remote Assistance/Access” service. The settings and state of this service are accessible from 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.

Improvements :

• Improved handling of SIP call redirection.

Fixed issues :

• Fixed issues related to the AARC remote control protocol.
• Issue for controlling from ScoopManager using a USB/Ethernet adapter: fixed.

Fixed issues :

• Double AoIP codec: connection icon missing on the root menu.
• Double AoIP codec: in some cases if two links are set and one is released, new incoming calls are rejected.
• Incorrect loading of a preset including “double SIP mono” active.
• If an incoming call is received while an AoIP session is running (single codec), call rejection with a “603 Decline” SIP code instead of “486 Busy Here”. Issue on version 1.07.01, fixed.

New features :

• Double AoIP codec: capability to operate as two mono AoIP codecs, each with its own SIP registration. (This new feature is available as an option)

Improvements :

• AoIP transmission statistics displayed on the “Connections” page of the html interface.
• The Scoop5 S-IP now reboots right after a “Factory Reset” is performed.
• More explicit error messages when an outgoing call fails.
• Removed “High temperature” from the “Alarm” html page.

Fixed issues :

• Inaccurate overload indication on html page.

Fixed issues :

• 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.

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.
• Instability of the synchronisation from the AES input: 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.
• Undesired interaction of the audio overload detection with the alarm indicators and relays.

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 the Ethernet interface.
• Web interface: added a title in the help section to remind the active tab.
• GPI management: ability to select “Relay Transmission” or “Call Profile” for each GPI.
• Ability to release a running connection by applying a pulse on the input loop (if “Loop Control” is not activated).

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.
• After a “Factory Reset”, the PLL was not correctly set.
• For the LL mode with MPEG L2 algorithm, “Error Correction” was not applied and was always set to “Mode 0”: setting now active.
• Relay transmission in LL mode with MPEGL2: unreliable state for the relay outputs when the connection starts up.

New features :

• Type of NAT displayed when STUN is active (“Status” page of html interface).
• Ability to change the Opus bitrate during transmission via the embedded html pages.

Improvements :

• 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

First released

Not released