Instreamer v3.17

http://www.barix.com/downloads/Instreamer_Standard_Firmware/1981/
- Mic audio input allows the FW to run on HW with Microphone input (e.g. Annuncicom)
- Use of DNS name in Streaming destination table (previously only numeric IP address)
- Support for Cisco Call Manager for Music on Hold applications

Streaming Client v2.24

http://www.barix.com/downloads/Streaming_Client_Firmware/1381/
- Triggered playback via metadata for Advert insertion controlled by the Broadcaster
- ASCII discovery for integration with ICgraph PC application
- Line-in support for the new Exstreamer 205, for local playback

SIP Client v1.12

http://www.barix.com/downloads/SIP_Client_Application/2331/
- Multicast socket for priority audio notification messaging, used for emergency and evacuation systems.

STL v2.12

http://www.barix.com/downloads/Studio_Transmitter_Link_STL_Application/2321/
- Mic audio input support for Annuncicom family devices
- SPDIF support for Instreamer HW
- Contact Input and Relay Output extension for Annuncicm1000

The introduction of a SIP gateway is the most effective way to merge audio over IP distribution with an already established SIP-based phone system. This is most useful when you are directing background music, paging and public address to loud-speaker systems as opposed to simply routing phone calls.

IT Telephony Article

All,

there is a new application firmware out for the Barix IP Audio devices. SIP Client version v01.07 brings a major change intended to make the use of the devices much more easy.
The Web UI, through which the device is configured, now offers “profiles”, to allow easy configuration of different use cases

1) SIP phone
This profile is intended for applications on bidirectional devices, for applications which are similar or identical to a SIP phone. The Barix devices can be used ideally in call stations, help points etc, preconfigured numbers can be called using the contact closure inputs etc.

2) SIP paging endpoint with Background Music support
This profile works on all devices which provide output capability (all Exstreamer and Annuncicom devices) and is typically used for “IP Speaker” application or to drive a paging amplifier.

3) SIP door station
This profile enables typical door station functionality, is able to operate in automatic “half duplex” mode to support simple door intercom panels (Aiphone), and also allows to activate a relay via SIP Info/DTMF to open the door.

4) SIP gateway
An interesting “glueware” application, when this is selected, the device can work as a “SIP to standard Multicast RTP” converter, will serve as an endpoint, and re-broadcast (or multicast) the received audio stream so it can be used with any device which can play audio from a standard RTP stream.

5) SIP based audio monitoring
This mode is typically used for audio surveillance, covert operations, etc. In this mode, the device can establish a call silently to a configured destination if it picks up audio above a certain level.

The SIP application firmware now supports the PS16, the desktop IP paging station from Barix, and on the Annuncicom 155, even the microphone and speaker supervision are supported.

The firmware can be downloaded, as usual, from our website free of charge and devices can be updated via browser.
Johannes

All,

the Barix Exstreamer P5 now starts shipping. This very affordable IP Audio device is actually .. an IP Speaker. And at the same time, it is not .. the loudspeaker is missing.

IP Speaker from Barix

So … why ? What is the reason behind this ?

As we all know, architects want to have the choice of design, form and color if it comes to selecting a speaker. Installers need to pick a speaker with the right environmental specs. And the price, or a specific manufacturer, may be a decision criteria too for the selection of the speaker.

Other manufacturers offer “IP Speakers”. Such an IP Speaker limits the choices for the installer drastically, as the speaker design part is fixed, and eventually already installed speakers, which would be perfectly useable, need to be pulled out and replaced.

The Barix Exstreamer P5 solves this dilemma. It is EVERYTHING an IP Speaker needs to be, with the exception of .. the speaker ! You can use almost any commercially available loudspeaker, being it round, square, outdoor, indoor, brown, white, shiny, expensive or cheap … as long as it has 8 Ohm impedance (the vast majority is of that type) and it can accept min. 5W RMS driving power.

The installation is incredibly simpe – two loudspeaker wires are connected to the Exstreamer P5 (screw terminal block included), and the device is hooked up to the network a Cat5 cable (PoE providing switch, or use an in-line power supply). NO need to wire power and network separately to the device. Also, if space is a constraint, you can separate the speaker from the Exstreamer P5.
A lot of flexibility – and the good news is that the combo “Exstreamer P5 plus reasonsable ceiling speaker” costs less than the typical combined IP Speaker device !

This IP Speaker solution from Barix supports all the standard stuff others do – SIP, priority multicast stream etc, and in addition, our devices can also play MP3 or even AACplus streams, resulting in CD quality background music, if wanted.
And – last not least – the Barix Exstreamer P5 also has a local interface port for an affordable volume/source control. So you can even select a background music channel, set the volume – and of course, even if you turn down the volume and listen to background music, a priority page comes through at a predefined level.

Check this out now .. the Exstreamer P5 is the better IP Speaker ! Coming from a trusted brand in the IP Audio and VoIP space since 10 years .. Barix ! The same technology used in the Exstreamer P5 also runs evacuation systems in nuclear power plants and oil refineries, it is rock bottom stable, so there is no need to power cycle your ceiling mounted IP speakers or find that reboot button ever …

“The IP Speaker without the speaker”(TM), now from Barix.

Barix IP Audio devices are an excellent solution when it comes to special SIP interfacing applications. Being it an audio monitoring application, where a call should be placed if a microphone picks up sounds above a certain threshold, or a SIP door station – the standard SIP application, which can be loaded on all Annuncicom and Exstreamer devices, serves well.

But do you know that you can also use a Barix device as a SIP gateway to traditional, multizone paging systems ?

How do you then select a zone ?

The SIP firmware can control local relays on the device, if existing (Exstreamer 500, for example, offers 4 relays). These can be commanded in the SIP call by means of standard signalling (SIP Info and RFC2833). Many IP PBX’es also allow you to send specific commands when you initiate a call, this is the perfect time to select a zone automatically.

The Barix wiki contains a nice article describing the exact configuration necessary of an Asterisk instance to offer the luxury of direct zone selection by extension. If done this way, a user of the phone system can just dial one of a selection of extensions, and the Barix SIP Gateway (actually, an Exstreamer 500 or Annuncicom 1000 device) will select the specific zone(s) to page to automatically, based on the actual extension called.
Here is a link to the general overview of how this is done: SIP Paging

And here is the article detailing the bits and pieces on how to configure Asterisk to directly control the relays from SIP

Greetings !

Johannes

Many people use IP Phones at home. A very popular phone is the Siemens Gigaset product range, which provides analog line as well as VoIP/SIP connectivity. Models C470IP, C475IP, S675IP and S685IP are currently in the market.

But .. how do you connect a door intercom or door station to them ?

Fortunately, the latest SIP firmware for the Annuncicom makes a perfect companion for the Siemens Gigaset IP phones, it can talk with the Siemens phone via IP !

• A visitor, who presses the “ring” button on the phone, places a SIP call to the Siemens base. This can be done “peer-to-peer”, so it won’t consume an account entry, and it does not need a PBX or server to work.

• The DECT phones will ring, indicating a call from the IP address of the Annuncicom

• The user can answer the phone talk to the person in front of the door, and also open the door by typing a configurable number sequence on the keypad (for example, “11″).

• Even multiple Annuncicoms can be used in this fashion, for example, for front door and back door !

The solution works, of course, also with other SIP PBXes like Asterisk, but the beauty with the Siemens IP phones is that there is no servers, no PBX, only little configuration needed.

Contact us if you are interested in more details, or check out our wiki pages.

Johannes

Have you ever faced the situation that you want to use Multicast between subnets, but Routers don’t forward it OR the application/device generating the Multicast traffic is using a TTL of 1, so the blocks don’t get forwarded by the router?

Barix developed a “Multicast routing/tunneling” firmware for the Barionet which turns that device into a flexible, multi site multicast forwarder/router. Effectively, it bridges multiple multicast groups between multiple sites, and can also include single hosts. The functionality is independent on the actual protocols used with the Multicast, being it automation, IP Audio, Video, VoIP, SIP or RTP.

Here is the rough concept:

• A Barionet is installed in every subnet where multicast traffic needs to be pickeed up or delivered.
• The device(s) does “UDP” tunnelling to forward the Multicast packets to the other Barionet(s) in the other subnet(s). At the same time, they serve as a tunnel receiver/endpoint to receive encapsulated Multicasts from the other networks.
• Up to 8 independent multicasts (address/port) can be configured, and up to 8 destinations – either Barionets in other networks as tunnel endpoints, or hosts which then receive the packets unicast.
• The TTL field of the multicast blocks can be “ignored and set to configured value” or handled as usual (decrementing).
• Tunnelling can be configured to use “IP over IP” or plain UDP – both have advantages and disadvantages.
• Monitoring of the application is possible via SNMP.

The first customer uses the application to route VoIP/VHF radio traffic between multiple operations centers in a large company. However, applications can be found wherever multicast needs to be routed between subnets and routers, IT providers or policies prevent that.

Please contact Barix if you are interested in that solution!

Johannes

As you probably have seen, Barix is supporting SIP since about 15 months now. A complete SIP stack and application, which is provided in source form, is in the Barix ABCL kit.

While requirements for an Audio over IP and a Voice over IP (VoIP) system are typically quite different, the Barix solution can serve both!

The SIP application can be used for emergency call posts on Annuncicoms (full duplex communication, initiated by a button press from a user), but also for output only applications, such as Paging speakers or interfaces to existing PA systems. In that case, the application supports multicast and also Auto Answer.

For use in Parking Systems or door access control, the built-in relay of an Annuncicom can be controlled via SIP Info.

So – where comes Audio over IP in place ?

Supporting MP3, AACplus, Ogg Vorbis and WMA (select hardware), the devices provide excellent capabilities for IP Audio distribution.

We are working on integrating both the Audio and Intercom/Voice/Paging functionality into one single application, when available, this will allow you to
do IP Audio distribution in high quality (for example, in Hotels, Cruise ships etc), yet, at the same time, have the device register as a VoIP “phone extension” on an IP based phone system.

For the Audio over IP functionality, the high quality music encoding can be used, and the devices automatically switch over to VoIP codecs when such functionality is required.

An alternative is to use one Barix device as a SIP gateway, allowing easy paging zone selection via “DTMF” (actually, SIP Info): The advantage here is that only one extension is used, which could save license cost on the PBX, and also enable synchronous paging on multiple devices via Multicast, which is not supported by many VoIP PBX’es so far.

A third option, available now, is the use of Bell Commander from Acrovista , a partner. It provides the same functionality on a PC, with an extensive Bell Scheduling and PC based Paging solution.

If you have an application where you need to deliver an RTP stream to multiple devices and your network infrastructure or source does not support Multicast, the Barix RTP replicator software may be the right thing for you.
Intended for Barix Audio over IP solutions, but useable with about any device or application or data, this software runs on either a Barionet or an Annuncicom, receives audio on a socket, and distributes the RTP packets to a list of up to 100 destination address/port numbers. You can also send to broadcast or multicast addresses.

Run on the Barionet, the application can easily forward a 100kbps IP Audio stream at 30-50 blocks per second to more than 100 destinations, with a delay of less than 20ms ! Using Barix hardware for this purpose has a lot of benefits: PC-Free technology, well below 4W power consumption, no moving parts, no themral issues, operational within 5 seconds of power up.

Customers of Barix use that solution since years to generate thousands of RTP streams to distribute IP Audio streams in real time.

.. and if you need to broadcast RTP in a “remote” network, the solution can be used as well … just configure only one target, the broadcast address, install the device in the remote location, and stream RTP to its receiving port. The RTP stream will be rebroadcasted in the destination network (multicast supported too, of course).

Questions ? Contact us !
Johannes

Barix devices can be perfectly used for IP intercom and emergency call post purposes. Our free ICGraph application (download from the website) can serve well for a PC based central solution, and we also offer a hardware console now (made by our OEM, MS Neumann Elektronik, who builds fully certified evacuation and industrial/oil and gas intercom systems using our IPAM IP Audio module).

However, what do you do if you need to handle potentially hundreds or thousands of call stations ?

Well – our SIP firmware comes to the rescue ! It can be used on all our IP Audio devices and makes them compliant to the standard SIP functionality – but with a twist: the source code is open, so you can add special behavior to the devices (and we can also do for you !)

As the central management software/solution, you can use any SIP based PBX, such as (free) Asterisk. You can typically configure hunt groups, queues, nighttime calling plans etc in a PBX easily, and use standard IP phones as consoles. Why re-invent the wheel if this is all available?

Of course, you can monitor the whole system independently from the PBX, add emergency capabilities (“all call” without the use of the PBX), background music etc.

We are happy to help planning such a solution – applications are numerous, being it an IP based nurse call system, highway emergency call boxes, a PA and intercom system for public transport … or an IP Audio entertainment and communications system for a cruise ship.

Johannes