aixtream features

Inputs & Outputs

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Audio
Audio over IP
DTMF
File
Monitor
On-Demand
Referece
RTP
Test
URL

Audio I/O

The aixtream PIPE element “Audio I/O” supports all types of conventional audio interfaces for inputs and outputs. This includes analog, AES/EBU and MADI. The specific channels used can be selected for both input and output. This ensures that even if many channels are available (for example because MADI or multiple analog or AES/EBU interfaces are used), the user keeps full control of the signal flow. This also means that the different types of I/O can be combined in whatever way is necessary. axitream acts as a full input/output matrix with format conversion and audio gateway/bridge functionality. Accordingly, users have full control over the channel mapping in input and output.

Ferncast’s Virtual Ports build on and enhance this full matrix concept. With Virtual Ports, users can set up a virtual audio router within the aixtream GUI, which can be used for a variety of applications. For example, the same signal can be reused for multiple outputs and multiple channels coming in with one signal can be split and send towards different outputs. This way, your SIP call (see below) could be recorded, streamed to a CDN as an Icecast and also made available as an MPEG TS all at the same time.

B

AES67, Ravenna & Dante

All modern audio-over-IP (AoIP) inputs and outputs are supported. This includes not just the widespread AES67 (via Ethernet), but also Ravenna, LiveWire (the Telos variant) as well as Dante interfaces. If desired, our AES67 implementation can also be used in a compatability mode for use in a Dante network.

Within AES67 there is a composition of s. n. Spalte, da können wir was rausziehen.

LiveWire from Telos has migrated to LiveWire+ and is also fully AES67 compliant. Though aixtream will support any of your devices audio I/O, whether AES67 pure, Ravenna or LiveWire. On top, DANTE which is spread in sound studios and some broadcasters will be supported as well. E.g. …

Ferncast’s Virtual Ports build on and enhance this full matrix concept. With Virtual Ports, users can set up a virtual audio router within the aixtream GUI, which can be used for a variety of applications. For example, the same signal can be reused for multiple outputs and multiple channels coming in with one signal can be split and send towards different outputs. This way, your SIP call (see below) could be recorded, streamed to a CDN as an Icecast and also made available as an MPEG TS all at the same time.

B

SIP Communication

Ferncast has developed multiple features to support and simplify SIP communication. All varieties of different SIP applications can be handled with aixtream, from classic bidirectional SIP calls to automatic answering machines with complex automatization and behavior. Even unusual input/output scenarios using SIP can be configured, including SIP bridges for other audio transmission formats or simultaneous recording of a call. All other audio processing and enhancement features you find listed here can also be used with SIP. This variety of features ensure both reliable and swift connections.

aixtream supports multiple registered SIP accounts on the same device. All of these can be used for different calls and the same account can be used for multiple calls (if the used SIP server supports this as well).

Besides the general account configuration, aixtream SIP supports setup of:
– Automatic call accept
– Automatic call rejection for non-whitelisted callers
– NAT traversal via STUN server
– Registration interval
– Transport protocol used (UDP, TCP, TS)

B