Cost-Effective Voice and Video Conferencing for Every Telephone User in the Enterprise
Video Conferencing Integrated into Award-Winning AsteriskÂ®-based IP-PBX Appliance
Today most businesses recognize the advantages of video conferencing, but many do not implement such systems due to their high price tag. In contrast to other video conferencing solutions currently in the market, the Xorcom XV3000 MCU (Multipoint Control Unit) enables organizations to conduct high quality multi-point video conferencing at an affordable price.
The solution has been integrated seamlessly into Xorcomâ€™s award-winning Asterisk based business telephony solution. SIP (Session Initiation Protocol) signaling ensures compatibility with a wide variety of endpoints, including the leading IP-based video phones as well as soft phones, significantly reducing the total cost of the solution.
Key Features of the XV3000 MCU and IP-PBX
- High Definition (HD) Voice and VideoÂ – Supports resolutions of up to 720p and Wideband HD Voice for unmatched voice quality bridge and density.
- Unmatched Price and ValueÂ – Organizations that do not have video and voice conferencing solutions cannot afford not to have it!
- Simple SolutionÂ – Built to be installed and up and running within minutes in any IP network environment, using simple Web management
- Seamless InteroperabilityÂ – Built on a field-hardened media processing foundation; provides standard-based SIP protocol to connect different devices
- Smooth TranscodingÂ – Transcoding and transrating between different endpoints with unmatched media quality
- Intuitive Web ManagementÂ – Provides full control to all levels of users, from IT managers to any user in the organization
- Reliable, Flexible TelephonyÂ – Unique PBX design ensures the reliability of a proprietary telephony system, while providing the flexibility, scalability, easy integration and competitive pricing afforded by an open source platform
XV3000 MCU Product Specification Highlights
Able to connect different devices, from room systems to desktop video phones, to video clients running on smart phones or tablets. Transcodes the video conference providing the best available video conference quality to every participant.
Signaling – SIP RFC Compliancy
SIP (Session Initiation Protocol), according to the following RFCs: RFC3261, RFC4566, RFC2976, RFC3262, RFC3263, RFC3264, RFC4317, RFC3581, RFC3966, RFC4028, RFC2833, RFC3550, RFC3551, RFC 3951, RFC3952
Support for both voice-only bridge or video and voice conferencing capabilities
G711a, G711u, G729, G722.1, G723, iLBC
MPEG4, H263, H264
QCIF, CIF, VGA, 4CIF and HD (720p)
Great Performance for SMB*
- Normal Conference Room (CIF)
- 1 conference room instance with 15 participants/12 viewable, or
- 2 simultaneous conference rooms with 15 participants/6 viewable
- High Conference Room (VGA)
- 1 conference room instance with 15 participants/10 viewable
- HD Conference Room (720p)
- 1 conference room instance with 15 participants/5 viewable
- Voice-only Conference
- Up to 12 simultaneous conference rooms with 30 participants per room
Transmitted Bandwidth from MCU
- HD Participant: 2 Mbps HD (1280X720) 15 FPS
- VGA Participant: 1.2 Mbps VGA (640X480) 30 FPS (High Conference) / 15 FPS (HD Conference)
- CIF Participant: 320Kbps CIF (352X288) 24 FPS
Minimum Bandwidth to MCU
The minimum required bandwidth from connecting endpoints is as follows:
- HD Participant: 1.2 Mbps VGA (640×480)
- VGA Participant: 320 Kbps VGA (640×480)
- CIF Participant: 320 Kbps CIF (352×288)
NAT/PAT Traversal Capabilities
Network and Port Address Translation capabilities for firewall traversals
Built-in simple Web management software allows users to configure the system, provision conferences, and monitor and control conferences in real time.
Password-protected administration access for platform administration and conference management. Different roles are supported for platform management and conference management.
Multiple Language Support
The built-in Web-based localization tool enables easy addition of new languages and translation of any text field from English to any language (includes right-to-left language support).
Configurable Meeting Rooms
A conference meeting room can be provisioned at any time, and used at any time. The conference rooms can be configured as voiceonly or voice and video conference rooms.
Multiple Access Numbers
Supports multiple access numbers, which can be registered in a SIP network as an endpoint or SIP trunk.
Participants may be recognized by the system and automatically connected to a predefined bridge; useful when connecting multiple conference rooms to a preset configuration.
Conference rooms can be moderated by a leader or set as unmanaged, to be used as needed. Dashboard enables the moderator to do the following:
- Set the dominant speaker
- View participant media information
- Mute/Unmute Voice
- Mute/Unmute Video
- Disconnect a participant
- Mute/Unmute all participants
- Invite multiple participants (Dial Out)
- Change the default layout to Equal or Dominant-based layout
Dynamic Video Layouts
As video participants join/leave the conference, the layouts automatically adjust to provide the optimal coverage of all viewable participants.
Energetic Voice Detection â€“ Dominant-Based Layout
Participant placement within the dominant speaker layout is updated based on continual measurement of all active voice participants resulting in the determination of the top five active (energetic) participants within a conference.
Real-time video and audio statistics per conference or per participant and total incoming and outgoing bandwidth usage is displayed in the leader dashboard.
SIP Desktop Sharing
Proprietary SIP desktop sharing mechanism (check below for currentÂ list of compatible endpoints)
The 2U XV3000 MCU series is available in four configurations:
- XV3000/NU: no telephony module support
- XV3047 â€“ 1 PRI (E1 or T1) port
- XV3055 â€“ 2 PRI (E1 or T1) ports
- XV3056 â€“ 4 PRI (E1 or T1) ports
Models supporting telephony interfaces also include hardware echo cancellation for enhanced voice quality. Additional telephony ports can be supplied via USB-connectedÂ Astribanks.
- Max. Users: 1000
- Max. Concurrent Calls: 480
- Max. Analog Ports: 800 (with Astribank add-ons)
- Ethernet Port: 10/100/1000 Mb/s
- Hard Disk Drive: 250 GB
- RAID1Â â€“ dual hard drive
- Fan redundancy
- Rapid Recoveryâ„¢Â – external backup and restore
- Internal backup and restore
- Restore to factory default
- Rapid Tunnelingâ„¢Â – secure access for remote troubleshooting
- AsteriskÂ® version 1.8x
- FreePBXâ„¢ GUI
- Elastix â„¢Asterisk distribution
Interoperability testing has been performed with a variety of endpoints:
Smartphones & Tablets
iOS & AndroidÂ
- Vippie (requires registration)
- Polycom (no NAT support)
Desktop Video Phones
- Grandstream (GXV3175)
- Yealink (VP530)
- snom (snom300-SIP)
- Fanvil (D800 + SE780)
- LG Ericsson IPECS (LIP8050)
Room System End Points
- Aver (HVC310 + HVC130) -Â supports desktop sharing
- Panasonic (KX-VC300)
- Xorcom XP0120; XP0100
- snom 300-SIP, v. 7.3.30 6057
* Performance may improve or degrade depending on endpoint capabilities.