Cisco SPA514G 4-Line IP Phone with 2-Port Gigabit Ethernet Switch, PoE, and LCD

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Brand Cisco
Data networking MAC address (IEEE 802.3)
IPv4 - Internet Protocol v4 (RFC 791)
ARP - Address Resolution Protocol
DNS - A record (RFC 1706), SRV record (RFC 2782)
DHCP Client - Dynamic Host Configuration Protocol (RFC 2131)
ICMP - Internet Control Message Protocol (RFC 792)
TCP - Transmission Control Protocol (RFC793)
UDP - User Datagram Protocol (RFC 768)
RTP - Real Time Protocol (RFC 1889) (RFC 1890)
RTCP - Real Time Control Protocol (RFC 1889)
DiffServ - Differentiated Services (RFC 2475)
ToS - Type of Service(RFC 791, 1349)
VLAN tagging 802.1p/Q - Layer 2 quality of service (QoS)
SNTP - Simple Network Time Protocol (RFC 2030)
Dimensions 0.42 x 8.35 x 1.73 in. (214 x 212 x 44 mm)
Features Full-featured 4-line business-class IP phone supporting Power over Ethernet (PoE)
Connects directly to an Internet telephone service provider or to an IP PBX
Dual Gigabit Ethernet switched ports, speakerphone, caller ID, call hold, conferencing, and more
Easy installation and secure remote provisioning, as well as menu-based and web-based configuration
Supports up to two Cisco SPA500S Expansion Modules adding up to 64 additional keys
Supports both SIP and SPCP Protocol with Cisco Unified Communication 500 Series
Indicator lights/LED Speakerphone on/off button with LED
Headset on/off button with LED
Mute button with LED
Message waiting indicator LED
Voicemail message retrieval button
Hold button
LED test function
Physical interfaces Two 10/100/1000BASE-T RJ-45 Ethernet ports (IEEE 802.3)
Handset: RJ-9 connector
Built-in speakerphone and microphone
Headset 2.5-mm port
Power supply Power supply is optional and is purchased separately
Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU
Switching type (100-240V) automatic
DC input voltage: +5 VDC at 2.0A maximum
Power adapter: 100-240V50-60 Hz (26-34 VA) AC input
Provisioning, administration, and maintenance Integrated web server provides web-based administration and configuration
Telephone keypad configuration via display menu/navigation
Automated provisioning and upgrade via HTTPS, HTTP, TFTP
TR-69, TR-104, and TR-111 Provisioning
Asynchronous notification of upgrade availability via NOTIFY
Nonintrusive in-service upgrades
Report generation and event logging
Statistics transmitted in BYE message
Debug and syslog server records: configurable per line
Voice gateway SIP v2 - Session Initiation Protocol version 2 (RFC 3261, 3262, 3263, 3264)
SPCP - Smart Phone Control Protocol with UC500
SIP proxy redundancy - dynamic via DNS SRV, A records
Reregistration with primary SIP proxy server
SIP support in NAT networks (including STUN)
SIPFrag (RFC 3420)
Secure (encrypted) calling via SRTP
Codec name assignment
Voice algorithms:
G.711 (A-law and µ-law)
G.726 (16/24/32/40 kbps)
G.729 A
Dynamic payload support
Adjustable audio frames per packet
DTMF - Dual-tone multifrequency, in-band and out-of-band (RFC 2833) (SIP INFO)
Flexible dial plan support with interdigit timers
IP address/URI dialing support
Call progress tone generation
Jitter buffer: adaptive
Frame loss concealment
VAD - Voice activity detection with silence suppression
Attenuation/gain adjustments
MWI - Message waiting indicator tones
VMWI - Voicemail waiting indicator, via NOTIFY, SUBSCRIBE
Caller ID support (name and number)
Third-party call control (RFC 3725)


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