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Shenzhen DBL Technology Limited

1 Channels VoIP Phone With SIP&H. 323, VoIP SIP Phone, IP SIP Phone manufacturer / supplier in China, offering 1 Channels VoIP Phone, IP Phone With SIP&H. 323 (EP-636), DBL 1/2/4/8-FXS VoIP Gateway (HT-882), 8 Port/Channel GSM VoIP Gateway with SIP&H. 323 (GoIP8) and so on.

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Supplier Homepage Product VoIP Telephone 1 Channels VoIP Phone, IP Phone With SIP&H. 323 (EP-636)

1 Channels VoIP Phone, IP Phone With SIP&H. 323 (EP-636)

FOB Price: US $40 / Piece
Min. Order: 1 Piece
Min. Order FOB Price
1 Piece US $40/ Piece
Get Latest Price
Production Capacity: 10000 Unit/Units / Month
Payment Terms: T/T, Western Union, Paypal, Money Gram
Basic Info
  • Model NO.: EP-636
  • Protocol: H.323&SIP
  • Trademark: DBL
  • Type: IP Phone
  • Support: Meeting
  • Origin: China
Product Description
The 1 channels VoIP phone SIP2.0&H. 323 (EP-636) is a quality phone with lots of features for both business and residential users. Its slim and upright design makes it an ideal desktop phone. The phone is based on ITU-H. 323 V4 and IETF SIP V2 open standards. The two protocols approach makes the phone to be compatible to most VoIP systems in deployment today. The Phone is designed for the ease of installation and setup. The PoE option simplify the installation in a PoE LAN environment. In addition, the second Ethernet Port allows the existing PC to be connected to the phone directly without addition an additional Ethernet Hub or Switch. Various configuration modes allow the user / system administrator to configure the phone automatically or quickly.

Key Features
Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)
All standard PBX functions
Four call appearances support two simultaneous calls
Two 10/100 Ethernet circuits connect to the LAN and an additional device
3-Line LCD (Icons, Alphabets/Numbers, Numbers)
Buttons and keys for all commonly used functions
Message waiting indicator
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Full duplex speaker phone
VLAN and QoS support
NAT Transversal and router functions
Power over Ethernet (PoE) or AC/DC adapter
Menu, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Phone Features
Call forward
Call transfer
Call hold
Mute
Redial
Display caller ID
Display call duration
Display date and time
Access voice mail
Send DTMF tones
Message waiting indication (MWI)
100 phone book entries
30 most recent call records for dialed, incoming, and missed calls
Adjustment of LCD contrast (4 levels)
Adjustment of handset volume (6 levels)
Adjustment of speaker phone volume (6 levels)
Enhanced Features
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese
Supported Standards
ITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450
RFC 1889 RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP "Replaces" Header
RFC 3892 SIP Referred-By Mechanism
Draft-ietf-sipping-CC-transfer-04 Session Initiation Protocol Call Control Transfer
Codec: G. 711 (A/µ Law), GSM, G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
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