VoIP Sip Adapter (PAP2T)

Product Description

Linksys Phone Adapter--PAP2T
Internet Phone Adapter with 2 Ports for Voice-over-IP
Feature-Rich VoIP Service through your High-Speed Internet Connection

Description:
The Linksys Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analog phones or use one of the ports for a fax machine. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and a reliable fax connection, even while using the Internet at the same time.

With Internet telephony, along with low domestic and international phone rates, an impressive array of special telephone features are available. Choose your preferred free local dialing area code, regardless of where you live. Or add a virtual telephone number in any area code, forwarded to your Internet phone. You can even add a toll-free number. The Linksys Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephony service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and much more.
Data Networking

MAC Address (IEEE 802.3)
IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883)
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) (RFC 1890)
RTCP - Real Time Control Protocol (RFC 1889)
Diffserv (RFC 2475), Type of Service - TOS (RFC 791/1394)
SNTP - Simple Network time protocol (RFC 2030)
Voice Gateway
SIPv2: Session Initiation Protocol v2 (RFC 3261, 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 μ -law),
G. 726 (16/24/32/40 kbps),
G. 729 A,
G. 723.1 (6.3 kbps, 5.3 kbps)
Dynamic Payload
Adjustable Audio Frames per Packet
Fax Capability
Fax Tone Detection and Pass-Through (Using. 711)
DTMF: In-band & Out-of-band (RFC 2833) (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 with Silence Suppression
Attenuation / Gain Adjustments
Flash Hook Timer
MWI - Message Waiting Indicator Tones
VMWI - Visual Message Waiting Indicator 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
Provisioning, Administration & Maintenance:
Web Browser Administration & Configuration via Integrated Web Server
Telephone Key Pad Configuration with Interactive Voice Prompts
Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP
Asynchronous Notification of Upgrade Availability via SIP NOTIFY
Non-intrusive, In-Service Upgrades
Report Generation & Event Logging
Stats in BYE Message
Syslog & Debug Server Records - Per Line Configurable

Physical Interfaces:
1 10baseT RJ-45 Ethernet Port (IEEE 802.3)
2 RJ-11 FXS Phone Ports - For Analog Circuit Telephone Device (Tip/Ring)
Indicator Lights/LED:
Phone1, Phone2, Internet, Power