Product Details

4 lines VoIP phone,support H323,SIP,VLAN and QoS

Place of Origin Guangdong, China
Brand Name SKYLINE
Model Number EP-8201
Type VoIP Phone

Product Features

Specifications

1 support ITU H323V4 and IETF SIPV2
2 ensure high quality speech
3 support VLAN and QoS
4 NAT Transversal
5 Router

Product Introduction:

The Sip Phone with H.323 and Sip (EP-8201) (EP-8201) 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 

 

Product Features:

  • Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
  • Single or Multiple Server Registrations
  • Two 10/100 Ethernet circuits connect to the LAN and an additional device
  • GSM module for making GSM calls
  • Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
  • VLAN and QoS support
  • NAT Transversal and Router functions
  • Voice prompts, HTTP Web, Auto Provision support for configuration and updates
  • Highly stable embedded Linux operating system in high performance ARM 9 Processor
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    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
  •  

     4 lines VoIP phone,support H323,SIP,VLAN and QoS4 lines VoIP phone,support H323,SIP,VLAN and QoS

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