ProVuCommunications LinksysSPA2100

KeyFeatures

Supports2SIPLines LANandWANConnection MessageWaitingIndication Distinctiveandconfigurableringing SpeedDialling DoNotDisturb CallWaiting CallerID CallTransfer ThreeWayConferenceCalling Downloadablefirmwareupdates
TheLinksysSPA2100hasconnectionsfor2 telephones that will operate with two SIP accounts therefore are independently configurable. This unit is ideal for users of Ethernet connectionservices,suchascablemodems. AswellasprovidingvoiceoverIPservices, this unit will allow you to connect multiple PCsaninstanthomenetwork. TheSPA2100willholdthepublicIPaddress providedbyyourISP.ThenusingaNATthis issharedbetweenthePC'sonyourLAN,but without the NAT getting in the way of the VOIPsessions. TheSPA2100isflexibleofferingdualFXS connections for analog handset and FAX plusdual Ethernetportsforconnectivityto theLANandBroadbandnetwork.

SPA Model

Service Lines
Active Calls 3 Way Conference

PSTN (FXO) Connection

SPA-2100 SPA-200 SPA-1001 SPA-3000
ProVuCommunicationsLtd,BankHouse,4PeelStreet,Marsden,HuddersfieldHD76BR Phone:01484840048Fax:01484840050Sip:provu@provu.co.uk http://www.provu.co.uk/Email:contact@provu.co.uk
Technical Data ProVu Communications Linksys SPA-2100
Telephony Features 2 voice port (RJ-11) fone LAN one WAN Imedance agnostics 8 Configurable settings Call Waiting, cancel call waiting, call waiting Caller ID Caller ID with Name/number (multi-national variants) Caller ID blocking Call forwarding:no answer,busy, all Do not disturb Call transfer Message waiting indication visual and tone based Call return Call back on busy Call blocking with tol restriction Delayed disconnect distinctive Ringing Calling and called number Off-hook warning tone Selective/Anonymous call rejection Hot line and warm line calling Speed dialling of 8 numbers/addresses Music on hold Package contents SPA-2100 phone adapter unit Power adapter RJ-45 Ethernet cable Quick installation guide Specifications Data networking MAC address (IEEE 802.3) Ipv4 Internet protocol v 4 (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 (RFC UDP User Data 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 791/1349) 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 algiotherms G.711 (A-law and u-law) G.726 (16/24/32/40 kbps) G.729 A G.723.1 (63 kbps, 5.3 kbps) Dynamic Payload Adjustable audio frames per packet Call Capability Fax tone detection pass-through -using G.711 DTMF:In band and out of band (RFC 2833) (SIP Info) Flexible dial plan support with inter digit times and Ip dialling 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 via FSK Polarity Control Hook flash event signalling Caller ID generation (Name and 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 Provision/configuration/authentication: HTTPS with factory installed client certificate HTTP digest Encrypted authentication via MD5 (RFC 1321) Up to 256-bit AES Encryption Provisioning, administration and maintenance Web browser administration and configuration via integral web server Telephone key pad configuration with interactive voice prompts Automated provisioning and 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 recorded per line configurable Physical interface 1 RJ-45 Port- WAN Ethernet 10base Interface 1 RJ-45 Port- Lan Ethernet 10base Interface 2 RJ-11 FXS Ports- For analog circuit telephone device Interface circuit (SLIC) Ring voltage:40-55 VRMS configurable Ring frequency: 10 Hz 40Hz Ring waveform: Trapezoidal andsinusoidal maximum ringer load: 3 REN On-hook/off hook characteristics: On-hook voltage (tip/ring):-50 V NOMINAL Off-hook current: 25 mA min Terminating imedance: 8 configurable settings including North America 600 ohms, European CTR21 Regulatory Compliance FCC (part 15, class B), cUL, CE, IC-003, A-tick Power supply DC input voltage:+VDC at 2.0 A Max Power consumption: 5 watts Switching type (100-240v) Automatic Power adapter: 100-240v 50-60Hz (26-34VA) AC input, 1.8m cord Indicator lights/LED LAN Activity/Link LED- on each Ethernet Port Status LED (In-Use, Provisioning, Idle, Alert) Amber Phone1, Phone2 LED (In-Use, Registered, Alert) Documentation Quick-Start installation and configuration guide User guide Dimensions 102mm x 124mm x 28 mm Unit weight 490g/ 1.08oz Operating temp 32 to 113F (0 to 45C) Storage Temp. -77 to 158F (-25 to 45C) Operating humidity 10 to 90% non condensing Storage humidity 10to90%noncondensing000

Sending & Receiving Faxes Through Linksys VoIP Adaptors Reliably transmitting faxes using Voice-over-IP is difficult to achieve. The problem lies in the fact that both the sending & receiving fax machines need to run pretty much perfectly in time with each other, also the data sent to the receiving fax must arrive in the same order as it left the sending fax. Over an analogue (PSTN) telephone line this is fine as variations in timing between sections of data (known as jitter) is pretty much zero, it is also trivial over an ISDN line as there is very strict timing & loss control. In both these examples data (including voice) is transmitted in an orderly fashion & arrives at the destination with the same order & timing as it was sent with. However, IP (Internet Protocol) was designed to carry non-time critical data such as web pages where it doesn't really matter what order the sections of data (called packets) arrive & if a packet is dropped along the way, it can be re-requested without causing any problems. IP communications are also subject to high jitter caused by network congestion, timing drift or route changes. As soon as you try to send an analogue fax (or any type of modem communication such as a credit card payment machine) over IP these effects are disastrous. Additionally to improve voice quality over IP, methods such as echo cancellation, variable jitter buffers & silence suppression can be employed. Any one of these will cause fax to fail. Sending Fax Using a Voice Codec One way of sending faxes using VoIP with Linksys adaptors is to use the common g.711a codec (A-LAW) which is normally used to transmit voice. This codec (including IP overheads) requires roughly 100kbps in either direction to work. In order to have a better chance of success the fax machine should be set to 14,400bps as any higher is unlikely to work over any VoIP connection. The entire range of Linksys adaptors support fax pass-through using the g.711a codec. It is necessary to alter a few settings in the line tab of the web interface of the Linksys adaptor. You only need to alter the line the fax will be plugged into. The following settings will give the best results:
Network jitter buffer: very high Jitter buffer adjustment: disable Call Waiting: no 3 Way Calling: no Echo Canceller: no Silence suppression: no Preferred Codec: G711a Use pref. codec only: yes
There is no real way to guarantee this method of faxing is going to work all the time. A lot will depend on the quality of your Internet service from your ISP. As always with Voice-over-IP you will see more reliable results when using a router with Quality-of-Service functionality to prioritise voice traffic over data traffic on your Internet link. Sending Fax Using the T.38 Protocol T.38 is the real-time Fax-over-IP protocol. Rather than just trying to use a voice codec to transmit an analogue fax signal, t.38 has been developed purely to allow fax or modem transmission over an IP network (such as the Internet). A gateway (such as a Linksys adaptor) that supports t.38 deals with all the delays, timing jitter, packet loss and isolates the analogue fax machine from these as far as possible. Also since t.38 sends faxes as image data rather than audio, the bandwidth requirements are half that of using g.711a (so roughly 50kbps in either direction). The t.38 protocol is still a new technology & the range of devices supporting it well is very limited. Currently only the Linksys SPA-2100 adaptor has t.38 support & you must be using firmware version 3.1.5 onwards. At the time of writing the SPA-2002 has sufficient memory capacity & processing power for t.38, the support is planned for a future firmware release from Linksys. Sending or receiving a fax using t.38 is automatic when using the SPA-2100. It is necessary to set the Fax Passthru Method to ReINVITE for the line the fax is connected to. The reliability of t.38 faxes should be much greater than using a voice codec but it is worth considering that the device you are sending a fax to or receiving a fax from, must also be a working t.38 device. If you are registered with a VoIP service provider who provide you with a PSTN phone number you want people to be able to fax to, you must check that your provider supports t.38.