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GXP 2120

SIP-T21E2

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Grandstream

Yealink

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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)

2 VoIP accounts, Call hold, mute, DND, One-touch speed dial, hotline, Call forward, call waiting, call transfer, Group listening, SMS, emergency call, Redial, call return, auto answer, Local 3-way conferencing, Direct IP call without SIP proxy, Ring tone selection/import/delete, Set date time manually or automatically, Dial plan, XML Browser, action URL/URI, Integrated Screenshots, RTCP-XR

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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

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDPMED, LDAP, TR-069, 802.1x, TLS, SRTP

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

SIP v1 (RFC2543), v2 (RFC3261), NAT transverse: STUN mode, Proxy mode and peer-to-peer SIP link mode, IP assignment: static/DHCP, HTTP/HTTPS web server, Time and date synchronization using SNTP, UDP/TCP/DNS-SRV(RFC 3263), QoS: 802.1p/Q tagging (VLAN), Layer 3 ToS
DSCP, SRTP for voice, Transport Layer Security (TLS), HTTPS certificate manager, AES encryption for configuration file, Digest authentication using MD5/MD5-sess, OpenVPN, IEEE802.1X, IPv6

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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

2 line keys with LED, 6 features keys: message, headset, redial, tran, mute, hands-free speakerphone, 6 navigation keys, Volume control 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

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

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Support for G.711µ/a, G.722 (wide-band), G.723 (pending), G.726-32, G.729 A/B, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO)

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FCC: Part 15 (CFR 47) Class BCE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1

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2 Port, 4 Port, 8 Port

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DTMF detection, and fax tone detection

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AC Power Supply Range 100 – 240 VAC

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Call History – Call Detail Records and Call Event Logging, Call Recording

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For USB Jack