Version 1.2.0b46038-1.05 required on the ZMAN cards of the AES67/RAVENNA modules : download here.
New features :
Control module : in the “Misc” html page of the Control module, ability to act as a gateway for providing access to the codec modules using the SNMP and AARC protocols. (Note that SNMP traps are not relayed by the control module)
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.
Version 1.1.4b[…] required on the ZMAN cards of the AES67/RAVENNA modules : download here.
New features :
Codec modules: 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.
For the AES67 modules, support of ST2022-7 : secure the AES67 interface by using redundant links.
Support of the AARC protocol for remote controlling the Codec modules.
The AoIP quality statistics now remain displayed on the “Connections” html page of a Codec module 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”.
In the “Codecs” page of the html interface of the Control module, added the “Tab Title” of each Codec module within the title of its related group box;
Configuration capabilities added to the “AES67” html page of the Control module : “General Configuration” (Sampling rate and resolution, PTP domain), ST2022-7 redundancy mode, and ability to reboot the AES67 module.
Regression introduced in version 1.03 of the Codec modules : GPO2 does not follow the received “Info 2” if GPO1 is not assigned the “Info 1” state.
Support of the version 2 of the rear mount module, including 4 GPIO; the two additional GPIO are assigned to the second codec of the module in double AoIP/SIP mode.
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.
New “Redial/Release” function to control the state of an audio connection from a single GPI.