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132 * 58 graphic character dot matrix with backlight LCD, 2 SIP accounts, 3 way conference, Auto Answer, Black list, Call Center Headsets mode, Call forward, call waiting, call transfer, Call Hold, Caller ID, Dialed/Received/Missed call (each 200), Direct IP call without SIP proxy, Hot Line, Local Phonebook (upto 1000 entries), Mute,DND, Redial, Voice Mail, White list

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Anonymous call, anonymous call rejection, Call Conference, Call park, call pickup, Call Recording, DND&Call forward state synchronization, Group pickup, Intercom, paging, Music on hold

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3.2″ 462 x 278-pixel TFT LCD, Dual Gigabit Ethernet Port, PoE, WiFi, Up to 4 SIP accounts,3-way conference,Connected with up to 2x TFT LCD expansion modules (RET/AET)

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2 custom ring tone article, 8 own ring tone, 8 volume adjustable + Mute mode, Acoustic Echo Cancellation (AEC), Adaptive Jitter Buffer (AJB), Auto Gain Control (AGC), Comfort Noise Generation (CNG), Full-duplex hands-free speakerphone with AEC (Acoustic Echo Cancellation), Narrowband CODEC: G.711a/u, G.723.1, G.726-32K, G.729AB, Packet Loss Concealment (PLC), Voice Activity Detection (VAD), Wideband codec: G.722, L16,

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AES encryption for configuration file, and P2P mode, Digest authentication using MD5/MD5-sess, DTMF: In-band, HTTP / HTTPS web server, IEEE802.1X, LAN / PC: support bridge mode, LLDP, NAT transverse: STUN mode, Open VPN support, Package tracing export, Phone lock, RFC2833, SIP connection mode: Proxy mode, SIP INFO, Support DNS SRV (RFC3263), Support QoS, Syslog, TLS, User and administrator level access control, VLAN

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Atcom

Grandstream

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GXP 2120

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Intuitive graphic user interface (GUI), downloadable phone book (XML, LDAP), support for anonymous call using privacy header, MLS (multi language support)
Voice mail indicator, downloadable custom ring-tones, call hold, call transfer
(attended/blind), call forward, call waiting, caller ID, mute, redial, call log, caller ID
display or block, Do-Not-Disturb (DND) and volume control
Multi-party conferencing (up to 5), dial plan prefix, off-hook auto dial, auto answer, early dial and speed dial (on some models)

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Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols
Supports multiple SIP accounts
Supports SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856, 3863) for use of 7 MFKs, SIP Dialog package (RFC 4235)
Supports SIP MESSAGE method (RFC 3428)
Stores up to 100 incoming IM messages

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DIGEST authentication and encryption using MD5 and MD5-sess, SRTP, SIP over
TLS (pending)

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6 line keys with dual color LED and 6 independent SIP accounts, HOLD, TRANSFER, CONF, VOLUME, HEADSET, MUTE, SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE, 3 XML
Programmable Softkeys, 5 Navigation keys

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Full-duplex hands-free speakerphone, headset enabled
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723, 1 (5.3/6.3K), G.729A/B, G.711 a/µ-law, G.726-32, G.722 (wideband), GSM and iLBC codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise
generation), AGC (automatic gain control)
Acoustic Echo Cancellation (AEC) with Automatic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control and packet delay & loss concealment

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English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, simplified and traditional Chinese, Korean, Japanese and more

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Via keypad/LCD, Web browser, or secure (AES encrypted) central
configuration file, manual or dynamic host configuration protocol (DHCP)
network setup
Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 ToS