VoIP Phone:
EP-8201(four lines IP Phone)
EP-636 (one line IP Phone)
Key Features
Open Standard VoIP Protocols ( 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
Graphical LCD
Full featured and programmable keypad for all phone functions
Phone display in English and Chinese (Other languages available upon request)
Buttons and keys for all commonly used functions
Message waiting LED
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
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), GSM, G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
EP-8201(four lines IP Phone)
EP-636 (one line IP Phone)
Key Features
Open Standard VoIP Protocols ( 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
Graphical LCD
Full featured and programmable keypad for all phone functions
Phone display in English and Chinese (Other languages available upon request)
Buttons and keys for all commonly used functions
Message waiting LED
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
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), GSM, G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
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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 |
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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), GSM, G. 729A/B, G. 723.1 DTMF: RFC 2833, In-band DTMF, SIP INFO |
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Operating temperature: 10° C to 40° C (50° F to 104° F) |
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Storage temperature: 0° C to 50° C (32° F to 122° F) |