uCLS1012A-VoIP

Voice and Video Communications

The uCLS1012A-VoIP combines the power of an Arm v8, 64-bit processor with SIPxtream communications and Mbarx Secure IoT firmware.

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Hardened communications

SIPxtream firmware doesn’t just enable voice and video communications it provides extensions to secure it, improve resilience, provide redundancy, monitor quality, improve intelligibility and even self-test.

Beyond the call

The uCLS1012A-VoIP supports up to 2 channels of VoIP, multicast PA, messaging, fileplay announcements and optional camera integration for streaming video. If your real-time communications application needs a workhorse, the uCLS1012A-VoIP answers the call.

Management built-in

We built the uCLS1012A-VoIP as a versatile communications platform, then we extended it to give you a full-featured secure WebUI, secure host protocol, facilities for local application integration, remote management and telemetry.

Edge software that gives you the edge

Standard Linux platform support in Buildroot, standard Docker container support and with SIPxtream and Mbarx firmware – unleash your next edge application with the uCLS1012A-VoIP.

Applications

Highly flexible, real-time communications for hardened devices.

Intercom Systems

Public safety intercoms and help points
Building entrance and access control systems
Elevator cabin, stairwell and parking garage communications
Passenger assistance communications
Patient care and nurse call systems

Public Address and Notification

School bells and communications systems
Campus wide PA paging systems
Fire panels and public safety communication
Automated evacuation messaging and instruction
Passenger information systems
Mass transportation, station and airport comms

Radio Systems and Specialized Applications

DMR/LMR radio convergence
Radio-Over-IP backhaul
Wi-Fi belt packs, headsets and wearables
Industrial equipment integration
Presentation and conference systems

Product Specifications

Easily integrated, Linux enabled with support and software stacks to get you to market.

Development Kit

uCLS1012A-VoIP module
Module host board, cables, power supply
Support site access (1 year)
Installation and maintenance support (1 year)
Documentation, integration schematics
SIPxtream firmware with VoIP, multicast PA, messaging, fileplay announcements and optional camera integration
Mbarx firmware with WebUI and remote management / telemetry
Mbarx System Manager tool (Win/Mac free trial)
Mbarx Virtual Control Panel tool (Win/Mac)

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Features

Hardware

Core Processor and Memory

NXP QorIQ^® LS1012A processor
800MHz 64-bit, Arm^® Cortex^®-A53 (Arm^® v8) core
512MByte DDR3-SDRAM
64MByte QuadSPI NOR flash (up to 128MByte)
32GByte eMMC (optional)

Connectivity and Peripherals

2x Gigabit Ethernet
Packet Forwarding Engine (PFE)
Security engine (SEC)
PCIe Gen 2
SATA Gen 3
USB 3.0 host
SDIO 3.0
SPI, UART, I2C, SAI/I2S
Up to 16 GPIO (1.1-3.6VDC tolerant)
Audio subsystem with class-D amplifier (optional)
802.15.4 radio with BLE/Thread^® stacks (optional)

Audio System

Up to 2x VoIP analog lines
1x full-duplex VoIP line with AEC and Noise Reduction
2x full-duplex VoIP lines with no AEC or Noise Reduction
1x full-duplex VoIP line with AEC and Noise Reduction and 1x half-duplex output with no AEC or Noise Reduction

Audio Connectivity

16-bit PCM with 8KHz, 16KHz sample rates
Headphones out – Mono, 1.25 Vrms (3.5Vp-p) 16-32Ω
Line out – 1.0 Vrms (2.8Vp-p) @ >10kΩ
Line input – 1.0 Vrms (2.8Vp-p) @ 5-15kΩ
Mic input – adjustable bias and gain 2.6Vdc 5-15 kΩ

Physical and Ratings

+3.3VDC +/- 5% 6W max (TBR)
-40°C to +85°C parts rating (TJ)
315-pin SMARC (MXM 3.0) module mating connector (MM70-314-310B1-2-R300)
82 x 50mm physical size

System Software

Overview

Linux OS with standard networking, DHCP, DNS, NTP…
Standard Buildroot platform support (downloadable) https://buildroot.org/
Preinstalled SIPxtream voice and video communications firmware
Preinstalled Mbarx Secure IoT endpoint firmware
Downloadable value-add software

SIPxtream Firmware

VoIP Call Control and Operation

Up to 2x analog audio VoIP lines (refer to audio system for supported configuration options)
SIP/RTP with full-duplex voice communication
SIP infrastructure (PBX) and P2P (standalone) compatibility
Call originate/receive, called-ID, hold/retrieve, transfer, auto-answer/hang-up, info messaging
Auto-answer ring timer
Covert listen mode
Suppress audible ring notification
Hotline (hook switch input) (optional)
Push-to-Call (PTC inputs) (optional)
PTT (optional)
Mbarx Secure IoT support with host protocol integration

VoIP Signaling and Media

SIP Fields (User, AuthID, CID, proxy, server, outbound proxy)
SIP, SDP, RTP, RTCP (UDP), SIP (TCP)
UAC/UAS stacks with standard transaction timers
SIP INVITE, ACK, BYE, CANCEL, REGISTER, INFO, OPTIONS, REFER, SUBSCRIBE / NOTIFY methods
REINVITE, REPLACES extension
OPTIONS ping request/response extensions
SIP priority, call type, answer mode extensions
Text messaging via SIP-INFO
Early media, late media
G.711ulaw, G.711alaw, G.722 (wide band) audio codecs, 20mS ptimes
iLBC, audio codec (optional), 20 or 30mS ptimes
DTMF RFC4733, in-band, SIP-INFO
DTMF tone generation
STUN, TURN/ICE
Configurable SIP tag lengths
SIP server via DHCP option 120 (optional)
SIP server via DNS-SRV (optional)

VoIP Calling Features

Caller-ID name, number, date, time, user provided privacy, block
Anonymous Call Rejection (ACR)
Message Waiting Indicator (MWI)
Call hold / retrieve
Music-on-Hold (MoH) compatibility
Held call ring-back
Call Waiting (CW), Cancel Call Waiting (CCW), disable call waiting
Call transfer – unattended, attended (REFER), attended (REPLACES)
Failed transfer ring-back (NOTIFY)
Missed call notification (when busy)
PA broadcast notification (when busy)
Do Not Disturb (DND)
Auto Call-Back on busy (ACB)
Call return
Local conference bridging (3WC) (optional)
Remote conferencing (meetME)
Phonebook / speed dials
Call forward
Outgoing call blocking rules
Door lock output signal (DTMF controlled)

Advanced VoIP Configuration

RTP port range
DiffServ RTP, SIP tagging
VLAN assignment
Runtime signalling overlays
Dynamic caller-ID
Dynamic registration, registration expiry offer period
Say IP address, SIP phone number

VoIP Security

sRTP
sRTCP
SIP-TLS
SIPs (security extensions)

Multicast PA Service (with control protocol)

Half-duplex, one-to-many, multicast RTP transmitter and receiver
SIP infrastructure (PBX) or P2P (standalone) compatibility
Up to 100 configurable talk groups/channels
Multicast Paging (MCPG) control protocol with keep-alive messages
Configurable protocol address / port
Unique multicast address per PA group
Protocol extensions (tones, text, security..)
Multiple concurrent transmitter support
Last-in wins arbitration (suitable for PTT/radio applications)
Stuck transmitter protection
Return to idle on loss of service
Recover session on acquisition of service
VoIP call notification during PA session
PA notification during VoIP call
VoIP (Rx)-to-MCPG (Tx) (suitable for wide-area paging)
G.711u (8KHz narrow band), G.722 (16KHz wide-band) audio codecs, 20mS ptimes
IGMP support
Compatible Linux PC application example (optional)

Generic Multicast PA Service (no control protocol)

Half-duplex multicast RTP receiver only
Up to 10 paging groups
Up to 10 priorities
One transmitter source
Unique multicast address per PA group
G.711u (8KHz narrow band) audio codec, 20mS ptimes
IGMP support
Compatible with common PBX intercom and PA support (e.g. Asterisk^®, Freeswitch^®, Polycom^®)
Compatible with stand alone PA and paging applications (e.g. Bell Commander^®)

Fileplay Service

Call progress announcements
Add/remove custom audio files
Dedicated local audio file storage (no external connection or streaming required)
Play audio file on trying / ringing, connect, terminate
Play audio file to analog, RTP or both
Scriptable (on-demand) audio file playback (suitable for station stop, passenger information or evacuation notification)
Basic IVR for system setup (e.g. load config file / reset)