Axis C8110 Network Audio Bridge
- Product Code: eq119121
-
Audio professionals looking for a versatile audio hub should look no further than the Axis C8110 Network Audio Bridge, a smart solution based on open standards for connecting and combining analog and digital audio systems.
Thanks to the built-in AXIS Audio Manager Edge application, you can utilize functions such as content scheduling and zone management in existing analog systems. With SIP support, it is possible to connect an IP telephony system for callouts in analog speakers. The analog audio inputs allow you to integrate your intercom system or music streaming box with network speakers from Axis.
Versatile Audio Hub Combines Analog and Network Audio I/O Ports for Buttons and Relays PoE for Easy Installation Based on Open Standards Easy Integration with Other Systems
Parameters | |
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Audio | Audio Management: Software Applications for Managing and Controlling Your Audio System Regardless of Its Size and Complexity: Included in the Product (Runs Inside the Device) Is Axis Audio Manager Edge, Offering Zone Management, Content Management, Content Scheduling, and Health Monitoring. Up to 200 Speakers and 20 Zones Are Supported. For Larger and More Advanced Systems Axis Recommends the AXIS Audio Manager Pro. Audio Encoding: AAC LC 8/16/32/48 kHz, G.711 PCM 8 kHz, G.726 ADPCM 8 kHz, Axis µ-law 16 kHz, WAV, MP3 in Mono/Stereo from 64 kb/s to 320 kb/s. Constant and Variable Bit Rate. Sampling Rate from 8 kHz up to 48 kHz. Audio Input/Output: Mic-In, Line-In, Line-Out Frequency Response: 20 Hz to 20 kHz Maximum Output Voltage: 0.91 Vrms Maximum Input Voltage: 202 Vrms |
Network | Security: Password Protection, IP Address Filtering, HTTPS Encryption, IEEE 802.1Xa Network Access Control, Digest Authentication, User Access Log, Centralized Certificate Management Supported Protocols: IPv4/v6b, HTTP, HTTPS, SSL/TLS, QoS Layer 3 DiffServ, FTP, SFTP, CIFS/SMB, SMTP, Bonjour, UPnPTM, SNMP v1/v2c/v3 (MIB-II), DNS, DynDNS, NTP, RTSP, RTP, TCP, UDP, IGMPv1/v2/v3, RTCP, ICMP, DHCP, ARP, SOCKS, SSH, NTCIP, SIP (Cisco, Avaya, Asterisk) |
System Integration | Application Programming Interface: Open API for Software Integration, Including VAPIX, One-Click Cloud Connection, Axis Camera Application Platform (ACAP). Support for Session Initiation Protocol (SIP) for Integration with Voice over IP (VoIP) Systems, Peer to Peer, or Integrated with SIP/PBX. Tested with Various Sip Client Such as Cisco, Bria, and Grandstream. Tested with Various PBX Suppliers such as Cisco and Asterisk. Audio Synchronization: Built-In Audio Synchronization for up to 30 Speakers with Unicast and Hundreds of Speakers with Multicast. No Additional Speaker Management Software or Hardware Required. Event Triggers: Call, Virtual Inputs, External Input, Axis Camera Application Platform (ACAP) Event Actions: Play Audio Clip, Send SNMP Trap, Status LED, File Upload Via Http, Network Share, and Email Notification via Email, HTTP, HTTPS, and TCP External Output Activation |
Casing | IP20-Rated Plastic and Metal Casing Color: Black NCS S 9000-N Memory: 512 MB RAM, 512 MB |
Power | 8-28 VDC, Max 4 W, Typical 1.85 W Power over Ethernet (PoE) IEEE 802.3af Type 1 Class 2 (Max 4 W) |
Connectors | RJ45 10BASE-T/100BASE-TX PoE I/O: 4-Pin 2.5 mm Terminal for Two Configurable Inputs/Outputs: Power: 2-Pin 5.0 mm Terminal Block DC Input Audio: 3 Pole 3.5 mm, RCA Line Out |
Storage | Support for MicroSD/MicroSDHC/MicroSDXC Card Maximum Size: 128 GB |
Reliability | Designed for 24/7 Operation |
Operating Conditions | Temperature: 4 to 122°F / -16 to 50°C Humidity: 10-85% RH (Condensing) |
Storage Conditions | -40 to 149°F / -40 to 65°C Humidity: 5-95% RH (Non-Condensing) |
Approvals | EMC: CISPR 35, CISPR 32 Class B, EAC, EN 55035, EN 55032 Class B, EN 61000-6-1, EN 61000-6-2, FCC Part 15 Subpart B Class B, ICES-3(B)/NMB-3(B), KC KN35, KN32 Class B, RCM AS/NZS CISPR 32 Class B, VCCI Class B Safety: UL 62368-1 Environment: IEC/EN 60529 IP20 |
Dimensions (H x W x D) | 0.9 x 5.0 x 2.2" / 2.4 x 12.7 x 5.6 cm |
Weight | 0.2 lb / 0.1 kg |
Packaging Info | |
Package Weight | 0.445 lb |
Box Dimensions (LxWxH) | 8.1 x 5.6 x 1.6" |
- Combines analog and network audio
- I/O ports for buttons and relays
- PoE for easy installation
- Based on open standards
- Easy integration with other systems
- When used with analog speakers, this versatile audio bridge adds IP functionality to your system, allowing you to convert analog speaker systems to IP. With AXIS C8110, you can connect any digital audio source such as 2N SIP Mic to your analog speaker system for live or pre-recorded voice messages.
- When used with network speakers from Axis, it allows for analog input from any audio source, converting the whole system and making it even smarter. A hybrid solution is also possible where you keep your analog system and add the benefits of network audio.
- Like all Axis products, AXIS C8110 is based on open standards (SIP, VAPIX, and ACAP). This means you’ll always have access to the latest applications being developed by our partners.
- With I/O, you can connect it to external buttons or relays to get extra functionality such as mute the system. And you can easily integrate AXIS C8110 with other systems. For example, with your network video system enabling announcements to be triggered by events captured on surveillance cameras, such as someone passing an exit the wrong way. You can also deliver voice messages by integrating it with any IP telephone system. And it connects to a standard network with Power over Ethernet.
- To get a microphone with the benefits of network audio, AXIS C8110 can be combined with AXIS TU1002-VE Microphone Kit. This combination lets you easily add intelligent analytics on the edge, for example detect aggressive behavior, broken glass, gunshots, and more.
With the built-in AXIS Audio Manager Edge application, you can efficiently manage and control your audio system. It enables functionalities such as zone management, content prioritization, and scheduling of content, so you can ensure the right content is sent to the right speakers at the right time. A single audio bridge is all you need for hundreds of speakers, making it a truly cost-efficient solution. This standalone, complete audio system offers an intelligent hub and allows analog audio sources to integrate with Axis network speakers. This means you can also play music from cell phones, tablets, computers, a professional music streaming box, and more.
Related Questions and Answers
Q: 1. Would I be able to adjust the volume from the android app?
A: Yes you can adjust the volume for the Bluesound Node using the BluOS Controller App on your Android device.
Q: 2. Does this work with clubhouse? i.e. 2x1 BT connected to POE switch onto Comp USB-C output to iPhone with Clubhouse on in addition to recording from said phone
A: I do not see this as a workable solution.
Q: 3. Current requirement for USB power source? 0.5A, 1A?
A: Audinate has not provided a spec for the amount of bus power used by the Dante AVIO 2x2 USB Type-A I/O Adapter for Dante Audio Network.
Q: 4. Can I use this device with a Shure MXA910 Ceiling Array Microphone?
A: Yes. According to the following page from Shure; "You can use a web application to control the MXA910 and Dante Controller to route audio. Download the Shure Web Device Discovery application to access your device’s web application." https://pubs.shure.com/guide/MXA910/en-US
Q: 5. Can this be used as a 'bi-directional analog audio" send and receive with a Dante system?
A: The Audinate Dante AVIO 2x2 AES3/EBU I/O Adapter for Dante Audio Network allows you to connect analog audio gear to a Dante network with two channels in and two channels out .
Q: 6. What is the maximum or recommended distance for deployments of this item?
A: Assuming a clear unblocked line of sight the manufacturer stated max range is 1640'.
Q: 7. 2 inputs x 1 output. Does this mean I will only have 1 mono signal output?
A: The Dante AVIO Bluetooth Adapter always outputs two Dante channels. It supports one A2DP connection for music and HFP connections for calls. For A2DP profile music, the output is stereo and uses both left and right channels. For HFP profile calls, the output is dual-mono, with the same content on both left and right channels.
Q: 8. Is there any one I can speak with about the bridge? A phone number?
A: You can contact B&H Pro Video via B&H Phone: 800.606.6969 or 212.444.6615 International phone number 001.212.444.6615., email: provideo@bhphoto.com.
Q: 9. Can I use this device to: 1) send analog out to ATEM Mic In. 2)USB to comp-1 streaming soft. 3) Dante to comp-2 running zoom?
A: Yes, the analog output from the Shure ANIUSB-MATRIX USB Audio Network Interface can be connected to the mic input on the Blackmagic Design ATEM Switcher (set the input level on the ATEM for line level audio signal). The USB port from the ANIUSB-MATRIX can be connected to a computer for streaming audio, and the Dante port on the ANIUSB-MATRIX can be connected to a network switch to receive Dante audio, Power over Ethernet (PoE), and data from the control software, then the second computer can be connected to the network switch to send and receive audio for the Zoom meeting.
Q: 10. is the Zen Stream power adaptor 9v or 12v?
A: According to page 01 in the manual for the iFi Zen Stream, the power adapter is DC 9V. https://www.bhphotovideo.com/lit_files/723691.pdf
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