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GXP 1450

GXP 1780

SIP-T29G

Grandstream

Grandstream

Yealink

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, re, ,
call log, caller ID display or block, Do-Not-Disturb (DND) and volume
control3-way conference,, plan prefix,, -plan support, off-hook auto, ,
auto answer, early, and speed

Hold, transfer, forward, 5-way conference, call park, call pickup, sharedcall-appearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook(XML, LDAP, up to 2000 items), call waiting, call log(up to 500 records), XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ringtones and music on hold, server redundancy and fail-over

16 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, 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, USB port (2.0 compliant) for: Bluetooth earphone through BT40, Contact synchronization through BT40, Wi-Fi through WF40, USB call recording through USB flash drive, Enhanced DSS Key

Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols, TR-069, 802.1x
Support multiple SIP accounts and up to 11 media channels
concurrently
Support SIP PUBLISH method (RFC 3903), SIP Presence package
(RFC 3856, 3863) for use of MFKs, SIP Dialog package (RFC 4235)
Support for SIP MESSAGE method (RFC 3428)

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPV6

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

User and administrator level passwords, MD5 and MD5-sess based
authentication, AES based secure configuration file, SRTP, TLS, 802.1x
media access contro

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, Kensington Security Slot (Kensington Lock) support

SIP v1 (RFC2543), v2 (RFC3261), Call server redundancy supported, NAT traversal: 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, LLDP/CDP/DHCP VLAN, ICE

2 line keys with dual-color LED and 2 independent SIP accounts, HOLD, TRANSFER, CONF, VOLUME, HEADSET, MUTE, SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE, 3 XML
Programmable Softkeys, 5 Navigation keys

8 line keys with up to 4 SIP accounts, 4 XML programmable context sensitive softkeys, 5 navigation/menu keys, 8 dedicated function keys for: PHONEBOOK, TRANSFER, CONFERENCE, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOLUME

10 line keys with LED, 10 line keys can be programmed up to 27 various features (3-page view), 8 features keys: message, headset, conference, hold, mute, transfer, redial, hands-free speakerphone, 4 context-sensitive “soft” keys, 6 navigation keys, Volume control keys, Illuminated headset key

Full-duplex hands-free speakerphone
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 (wide-band), 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), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control (patent-pending) and packet delay and loss
concealment
HD audio handset with HD wideband audio codecs for excellent doubletalk
performance

Yes, HD handset and speakerphone with support for wideband audio

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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

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

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

g Firmware upgrade via TFTP / HTTP / HTTPS, mass provisioning using TR-069
or AES encrypted XML configuration file

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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