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HDA50 Open SIP Headset Adapter

  • HDA50
  • HDA50
  • Hear and be heard, even in busy call centers
  • Take and make calls easily using most USB headsets
  • Track data and troubleshoot remotely
  • Save time, money and stress with auto updates
Data Sheet

Overview

Great news for soft phones

Your customers deserve great service. Your reps deserve great tools. Make both sides of the conversation happy with the HDA50 open-SIP headset adapter that delivers desk phone call quality when using a soft client. It’s ideal with Plantronics models but plays well with most USB softphone headsets, too. Remote data tracking and troubleshooting and set-and-forget software updates make it even easier to use.

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Benefits
  • Desk phone-like call reliability
  • Simple by design
  • Reduces support costs

Provides the optimal solution: the use of a soft client for call management and HDA50 for voice. Delivers desk phone call quality for assurance of maintaining critical conversations while using soft client for call management

Its space-saving, compact size allows for less desktop clutter and cost-effective shipment. Once the HDA50 is provisioned and deployed, CSRs just plug in their headset and start taking calls

Provisioning and deployment is simple and quick, which reduces time and cost for IT. While optimized for Plantronics headsets, the HDA50 also works with most other USB headsets, which can help lower overall costs.

Features

HDA50 Features

Specifications

  • PRODUCT REQUIREMENTS
    • Access to internet via a switched ethernet port
    • Active phone service subscription with all required SIP credentials to make and receive calls
    • USB Type-A headset supporting HID protocol
    • HDA50 VoIP endpoint
    • Power adapter
    • 1 x RJ-45 ethernet cable (80 inches/203 centimeters)
    • Quick start/installation guide
    • Velcro tape
    • 1 year
    • 6.9 cm x 6.9 cm x 3.0 cm (2.7 in x 2.7 in x 1.2 in)
    • 198 grams/7 ounces
    • Shipping weight: 340 grams/12 ounces (including power supply, ethernet cable and packaging)
    • Internet (WAN): 2 x 10/100BaseT ethernet port (802.3)
    • USB Type-A: USB 2.0 (headset port)
    • Reset button: located on bottom of case
    • LED indications: power on, status, ethernet activity (WAN), upgrade in progress status, packet RX/ TX and USB headset status
    • Phone: 2 x RJ-11 FXS analog phone port (optional)
    • Call routing rules
    • SIP service configurable inbound call routing rules
    • In-band DTMF (G.711)
    • Out-of-voice band DTMF (RFC 2833)
    • Out-of-voice band DTMF (SIP INFO method)
    • Call progress tone generation
    • Tone profile per SIP Service Provider and Polycom Device Management Service for Service Providers
    • Ring profile per SIP Service Provider and Polycom Device Management Service for Service Providers
    • Full duplex audio
    • G.165, 168 echo cancelation
    • VAD—voice activity detection
    • Silence suppression
    • Comfort noise generation
    • Three-way conference calling with local mixing
    • Daylight savings time support—worldwide
    • Call waiting
    • Maximum session control
    • Ethernet Authentication (IEEE 802.1X)
    • MAC address (IEEE 802.3)
    • UDP (RFC 768) in SSL/TLS
    • TCP (RFC 793) in SSL/TLS
    • IP version 4, IPv4 (RFC 791)—static IP and DHCP support
    • ICMP (RFC 792)
    • ARP—address resolution protocol
    • Domain name system (DNS) A records (RFC 1706) and SRV records (RFC 2782) RTP (RFC 1889, 1890), RFC 5966
    • RTP/RTCP (RFC 1889), DHCP client (RFC 2131)
    • DiffServ (RFC 2475)—independently configured: service, SIP and media
    • ToS (RFC 791, 1349)—independently configured: service, SIP and media
    • VLAN Tagging (802.1p—independently configured: service, SIP and media
    • SNTP (RFC 2030)—primary and secondary NTP servers
    • LLDP-MED
    • Traverse through web proxy serveraccess via HTTP, TFTP— HTTPS
    • Remote access interface: user name and password access via HTTP, TFTP— HTTPS
    • Device web page standard: HTTP v1.1, XML v1.0
    • Secure remote provisioning: HTTP, HTTPS
    • Simple Certificate Enrollment Protocol (SCEP)
    • Four (4) service provider configuration profile assignments (ITSP 1–4)
    • Four (4) service/trunk subscription profile assignments (SP 1–4)
    • SIPv2 (RFC 3261, 3262, 3263, 3264)
    • SIP over UDP
    • SIP over TCP
    • SIP over TLS
    • 4 SIP service provider service sessions—concurrent operation
    • 1 Polycom Device Management Service for Service Providers session
    • SIP proxy redundancy—local or DNS based SVR, primary and secondary fallback list restrict source IP address
    • Maximum number of sessions—independent per service
    • 4 trunk groups
    • Voice gateway—direct dialing
    • G.711 A-law (64 kbps)
    • G.711 μ-law (64 kbps)
    • G.726 (32 kbps) G.729a (8 kbps) iLBC (13.3, 15.2 kbps) codec pre-selection code
    • Wideband codec (USB headset): G.722 (64 kbps), Opus
    • Voice processing per SIP service—TX/ RX audio gain, echo cancellation
    • Adjustable audio frames per packet
    • Codec name assignment
    • Codec profile per SIP SP and Service Provider and Polycom Device Management Service for Service Providers
    • Dynamic audio payload
    • Packet loss concealment
    • Universal switching with fixed US, EU, UK style plug prongs (model dependent)
    • AC input: 100 to 240 volts 0.3A 50–60Hz (26–34 VA)
    • DC input: +12V 1.0 amp max
    • FCC part 15 class B
    • A-tick
    • CE
    • ICES-003
    • RoHS
    • WEEE
    • UL/cUL—power adapter
    • Operating temperature: 0º to 45º C (32º to 113º F)
    • Relative humidity: 10% to 90% non-condensing
    • Storage temperature: –25º to 85º C (–13º to 185º F)
    • Remote provisioning: XML via TFTP or HTTP, (TR069/TR104 parameter naming syntax)
    • Secure remote provisioning: HTTPS, encrypted XML via HTTP or TFTP— dedicated user name and password
    • Secure remote firmware update: encrypted binary file via TFTP or HTTP + dedicated user name and password
    • Customization: OBi-ZT: Obihai zero-touch automatic customization and configuration **
    • Call history (CDRs): call detail records on Service Provider and Polycom Device Management Service for Service Providers web page, export to XML
    • LED Indications: power, device status, upgrade progress status, ethernet activity, USB headset status
    • Session information: SIP session status, Polycom Device Management Service for Service Providers status
    • Primary SIP service set-up wizard: dedicated device web page for quick ITSP account set-up
    • System settings back-up/restore: save and restore configuration via XML file to/from a local folder
    • RTP transport type
    • Audio codec type (Tx/Rx)
    • Audio codec type (Tx/Rx)
    • RTP packetization—in multiples of 10ms (Tx/Rx)
    • RTP packet count (Tx/Rx)
    • RTP byte count (Tx/Rx)
    • RTP byte count (Tx/Rx)
    • Packets out-of-order
    • Packets interpolated
    • Packets late (dropped)
    • Packets lost
    • Packet loss rate %
    • Packet drop rate %
    • Jitter buffer length–ms
    • Received interarrival jitter–ms
    • Jitter buffer underruns
    • Jitter buffer overruns
    • Configurable call progress tone
    • Call progress tone profiles (2)
    • Dial tone
    • Busy tone
    • Ringback tone
    • Reorder tone
    • Confirmation tone
    • Holding tone
    • Second dial tone
    • Stutter tone
    • Howling tone
    • Prompt tone
    • Call forwarded tone
    • Conference tone
    • SIT tones (1–4)
    • Ringing and call waiting tone configuration
    • Ring patterns (10)—configurable
    • Call waiting tone patterns (10)—configurable
    • Call waiting tone pattern profiles (2)
Data Sheet

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