> Voice over Ip by Uyless D Black

model- 5801 vendor- zoom telephonics features- voip telephone adapter with g.711 ilbc & g.729 codecs does not include voip service. the economical zoom model 5801 and model 5802 voice over ip telephone adapters allow standard analog telephones to make and receive calls over a broadband internet connection. simply plug a conventional or cordless phone into the adapter and then plug the telephone adapter into a broadband- connected router or internet gateway. a minimum of five attached phones can be rung simultaneously using the built-in ring generator. the model 5801 and model 5802 support industry- standard session initiation protocol (sipv2) and work with a wide range of service providers and sip-based voip equipment. both models provide one fxs and one fxo port and allow calls to be bridged between the public switched telephone network (pstn) and voice over ip. the fxo port of the adapters also include the teleport an intelligent relay that allows a single phone to place and receive both voip calls and calls over the pstn. pstn fail-over allows the 5801 and 5802 to automatically route calls over the dial-up phone network when power is lost. pstn support also allows emergency dialing using services like 911 112 or 999 calling. voice over ip service providers can deliver a rich variety of advanced telephone services through the model 5801 or 5802 including class features such as call waiting caller identification call transfer call hold call forwarding distinctive ringing and voice message waiting indication. the voip terminal adapters can be configured remotely using a tftp or http download from the service provider and updates to the firmware can be automatically delivered. local configuration is done with a browser-based graphical user interface. includes voip telephone adapter power adapter quick start guide cd rom ethernet cable phone cord.* economical solution provides voice over the internet using regular telephones.* uses industry- standard session initiation protocol (sip).* clear natural-sounding voice quality.* provides one fxs and one fxo port* supports remote delivery of firmware updates* zooms teleport feature allows one or more analog phones to place and receive calls over the internet and also over the public switched telephone network.* supports automatic provisioning by service providers* supports quality of service (qos) * supports stun (simple traversal of udp over nats)* supports advanced telephone services including class features such as call waiting caller identification call transfer call hold call forwarding distinctive ring and voice message waiting indication.* proven compatibility with sip-standard servers from asterisk cisco netcentrex quintum and more. -- specifications ----------------- ----------------an alog telephone ports- (1) fxs type loop-start interface with rj-11 (1) fxo analog interface with teleport and rj-11 up to 5 ren (ringer equivalence number) supports more than 5 typical telephones. programmable ring patterns call progress tones supported: initial dial tone secondary dial tone stuttered dial tone message waiting dial tone call forward dial tone pre-ringback dial tone ring back tone call waiting tone call holding tone call disconnect tone call conference tone busy tone reorder tone (network busy/fast busy) off hook warning. power fail over auto switch to pstn for emergency calling using 911 and other programmed three digit codes. fxs to fxo call bridgingindicator s - leds for power lan link voip ready voip call in progress pstn port ready voice message waiting.voice over ip (voip) protocols- sipv2 - session initiation protocol (rfc 3261 3262 3263 3264). sdp - session description protocol (rfc 2327) rtp - real time protocol (rfc 1889 1890) rtcp - real-time control protocol (rfc 1889) x-nse - tone events for sip/rtp (rfc 2833) avt - tone events for sip/rtp (rfc 2833) power-on auto registration re-registration with sip proxy server sip over udp sip authentication (hhp digest with md5)n...