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Perfect Sip Adaptor From Sipcom
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Technical Specifications
Data Networking:
Mac Address(IEEE802.3)
IPv4-Internet Protocol Version 4(RFC 791)upgradeable to v6(RFC1883)
ARP-Address Resolution Protocol
DNS-A Record (RFC 1706) SRV Record(RFC 2782)
DHCP Client-Dynamic Host Configuration Protocol(RFC 2131)
ICMP-Internet Control Message Protocol(RFC792)
TCP-Transmission Control Protocol(RFC793)
UDP-User Datagram Protocol(RFC768)
RTP-Real Time Protocol(RFC 1889)(RFC1890)
RTCP-Real Time Control Protocol(RFC1889)
DiffServ(RFC 2475) Type of Service-TOS(RFC791/1349)
SNTP-Simple Network Time Protocol(RFC2030)
Voice Gateway:
SIPv2-Session Initiation Protocol Version 2(RFC3261,3262,3263,3264)
SIP Proxy Redundancy-Dynamic via DNS SRV,A Records
Re-registration with Primary SIP Proxy Server
SIP Support in Network Address Translation Networks-NAT(incl STUN)
Secure(Encrypted)Calling via Pre-Standard Implementation of Secure RTP
Codec Name Assignment
Voice Algorithms:
 G.711(A-law and mu-law)
 G.726(16/24/32/40 kbps)
 G.729 A
 G.723.1(6.3kbps,5.3kbps)
Dynamic Payload
Adjustable Audio Frames Per Packet
Fax Tone Detection Pass-Through
DTMF:In-band & Out-of-Band(RFC2833)(SIP INFO)
Flexible Dial Plan Support with Interdigit Timers and IP Dialing
Call Progress Tone Generation
Jitter Buffer-Adaptive
Frame Loss Concealment
Full Duplex Audio
Echo Cancellation-(G.165/G.168)
VAD-Voice Activity Detection W/Silence Suppression
Attenuation / Gain Adjustments
Flash Hook Timer
MWI-Message Waiting Indicator Tones
VMWI-Via FSK
Polarity Control
Hook Flash Event Signaling
Caller ID Generation(Name & Number)-Bellcore,DTMF,ETSI
Music on Hold Client
Streaming Audio Server-Up to 10 Sessions
Security:
Password Protected System Reset to Factory Default
Password Protected Admin and User Access Authority
Provisioning/Configuration/Authentication
HTTPS with Factory Installed Client Certificate
Sipcom SecureProvision
HTTP Digest-Encrypted Authentication via MD5(RFC1321)
Up to 256-bit AES or RC4 Encryption
Provisioning,Administration and Manintenance:
Web Browser Administration and Configuration via Integral Web Server
Telephone Key Pad Configuration with Interactive Voice Prompts
Automated Provisioning & Upgrade via HTTPS,HTTP,TFTP
Asynchronous Notification of Upgrade Availability via NOTIFY
Non-Intrusive,In-Service Upgrades
Report Generation and Event Logging
Syslog and Debug Server Records
Per Line and Purpose Configurable Syslog and Debug Options
Physical Interfaces:
One RJ-45 Port-Ethernet 10baseT Interface(IEEE 802.3)
One RJ-11 FXS Port-For Analog Circuit Telephone Device(Tip/Ring)

http://www.sipcom.com.cn/ata1000.html





_________________
Daniel Choi<br><br>E-mail(MSN):Daniel@sipcom.com.cn<br>Yahoo:Danielchoi79@yahoo.com<br>Website:www.sipcom.com.cn<br>Sipcom technology,inc.
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SIM Server Solutions by FL-Connect. Dramatically reduce SIM card loss!
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sipcom wrote:

Technical Specifications
Data Networking:
Mac Address(IEEE802.3)
IPv4-Internet Protocol Version 4(RFC 791)upgradeable to v6(RFC1883)
ARP-Address Resolution Protocol
DNS-A Record (RFC 1706) SRV Record(RFC 2782)
DHCP Client-Dynamic Host Configuration Protocol(RFC 2131)
ICMP-Internet Control Message Protocol(RFC792)
TCP-Transmission Control Protocol(RFC793)
UDP-User Datagram Protocol(RFC768)
RTP-Real Time Protocol(RFC 1889)(RFC1890)
RTCP-Real Time Control Protocol(RFC1889)
DiffServ(RFC 2475) Type of Service-TOS(RFC791/1349)
SNTP-Simple Network Time Protocol(RFC2030)
Voice Gateway:
SIPv2-Session Initiation Protocol Version 2(RFC3261,3262,3263,3264)
SIP Proxy Redundancy-Dynamic via DNS SRV,A Records
Re-registration with Primary SIP Proxy Server
SIP Support in Network Address Translation Networks-NAT(incl STUN)
Secure(Encrypted)Calling via Pre-Standard Implementation of Secure RTP
Codec Name Assignment
Voice Algorithms:
 G.711(A-law and mu-law)
 G.726(16/24/32/40 kbps)
 G.729 A
 G.723.1(6.3kbps,5.3kbps)
Dynamic Payload
Adjustable Audio Frames Per Packet
Fax Tone Detection Pass-Through
DTMF:In-band & Out-of-Band(RFC2833)(SIP INFO)
Flexible Dial Plan Support with Interdigit Timers and IP Dialing
Call Progress Tone Generation
Jitter Buffer-Adaptive
Frame Loss Concealment
Full Duplex Audio
Echo Cancellation-(G.165/G.168)
VAD-Voice Activity Detection W/Silence Suppression
Attenuation / Gain Adjustments
Flash Hook Timer
MWI-Message Waiting Indicator Tones
VMWI-Via FSK
Polarity Control
Hook Flash Event Signaling
Caller ID Generation(Name & Number)-Bellcore,DTMF,ETSI
Music on Hold Client
Streaming Audio Server-Up to 10 Sessions
Security:
Password Protected System Reset to Factory Default
Password Protected Admin and User Access Authority
Provisioning/Configuration/Authentication
HTTPS with Factory Installed Client Certificate
Sipcom SecureProvision
HTTP Digest-Encrypted Authentication via MD5(RFC1321)
Up to 256-bit AES or RC4 Encryption
Provisioning,Administration and Manintenance:
Web Browser Administration and Configuration via Integral Web Server
Telephone Key Pad Configuration with Interactive Voice Prompts
Automated Provisioning & Upgrade via HTTPS,HTTP,TFTP
Asynchronous Notification of Upgrade Availability via NOTIFY
Non-Intrusive,In-Service Upgrades
Report Generation and Event Logging
Syslog and Debug Server Records
Per Line and Purpose Configurable Syslog and Debug Options
Physical Interfaces:
One RJ-45 Port-Ethernet 10baseT Interface(IEEE 802.3)
One RJ-11 FXS Port-For Analog Circuit Telephone Device(Tip/Ring)

http://www.sipcom.com.cn/ata1000.html

send quotation


edygrois@gmail.com



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