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Packing Box Dimension :27.5*22.8*8.8cm G.W:1.05kg Carton Dimension:56.2*44.6*24.3cm 10pics/carton G.W:11.7kg

Specifications

free voip calls
1support ITU H323V4 and IETF SIPV2
2 ensure high quality speech
3 support VLAN and QoS
4 NAT Transversal

voip phone

Product Introduction:

The one channel VoIP Phone,IP Phone with Sip&H.323 (VP-202) is a new addition to our IP Phone family. It is designed as a low cost version of the model VP-202. The phone has a small alpha numeric LCD, 2 Ethernet ports, and a headset port. It has all the basic features available in a traditional phones and is intended as a basic model for VoIP deployment to replace the traditional telephone service. It supports both Standard SIP V2 and H.323 V4. Customization/ODM is welcome!

 

Basic Phone Features

Call forward

Call transfer

Call hold

Mute

Redial

Disay caller ID

Display call duration

Display date and time

SMS Capable

Access voice mail

Send DTMF tones

Message waiting indication (MWI)

100 phone book entries

30 most recent call records for dialled, 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), G.729A/B, G.723.1

DTMF: RFC 2833, In-band DTMF, SIP INFO