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The following is quoted from www.stikeman.com/newslett/TelJun04.htm.

Two key differences between VoIP and traditional telephone technology are the network over which a call is transmitted and the transmission format. In traditional telephony, a call over the public switched telephone network (PSTN) generally results in the creation of an end-to-end circuit that establishes a physical connection between the caller and the called party through the wires, cables and switches of the telephone network. The circuit exists for the duration of the call until hang-up, and is typically dedicated to the voice signals flowing between the two parties participating in the call. In contrast, in a VoIP call there is no dedicated circuit. The contents of a VoIP call flow between the caller and the called party over various networks that comprise the Internet. This happens through a router or switch operated by the VoIP provider, which matches a telephone number with an IP address. Different portions of a VoIP call may be routed over different transmission paths, with the contents of the call moving over the Internet amidst other traffic, including other VoIP calls, e-mails, miscellaneous data and video traffic.

The call format is the second feature distinguishing VoIP from traditional telephony. Voice communications on a PSTN call begin as analog signals, or electronic sound waves, which are converted into digital format (i.e., a series of ones and zeroes). The digits are transmitted in sequence over the circuit and converted back into analog format to be heard by the other party on the call. The digits move across the circuit in sequence, corresponding to the voice signals on the call. Spoken words on a VoIP call are converted from analog into digitized Internet Protocol (IP) packets. A packet is analogous to an envelope with data on the inside and sequence coding, as well as originating and destination addresses, on the outside. The VoIP users telephone is connected to a router. These packets enter the Internet at the router and are sent individually, though not necessarily in sequence, over the Internet. At the router closest to their destination (i.e. the other party on the call), they are reassembled in original sequence and converted back into analog format for reception by the recipient. A key point associated with VoIP is the significantly lower transmission costs of this technology.

Feedback received on this FAQ entry:
  • Try out Ozeki VoIP SIP SDK. You can download and use trial version free of charge. Here is the link: http://voip-sip-sdk.com/p_21-download-ozeki-voip-sip-sdk-voip.html I would like to mention another page as well. It is useful to get more information on IVR systems: http://www.voip-sip-sdk.com/p_90-how-to-handle-the-speaker-and-the-microphone-in-c-net-when-building-a-softphone-voip.html I think it is worth reading those valuable webpages. Have a nice day!

    2012-09-12 05:32:32 (38934683 See Profile)

by Styvas See Profile edited by canoe See Profile
last modified: 2005-06-16 22:38:12

An ATA is an Analog Telephony Adapter which interconnects your standard Telephone handset to a VoIP service; without the use of a computer. Primus TBB currently uses the DLINK DVG-1120M ATA Gateway which supports the MGCP VoIP protocol. They have recently introduced two Linksys ATAs (the SPA2100 and 2102) which support the SIP VoIP protocol.

A VoIP Gateway is a seemingly simple device that allows one to connect a standard telephone to one port, a broadband modem to another port, and a computer or network sharing device to yet another port for sharing of your broadband connection.

D-Link has this to say about their product:

The DVG-1120 is a VoIP Gateway equipped with two standard phone ports, one lifeline voice port for fail-over to standard phone services, one WAN Fast Ethernet 10/100BaseTX Port, and one LAN Fast Ethernet 10/100BaseTX Port.

With the integration of both voice and data, the DVG-1120 offers Small-to-Medium-Size Businesses (SMB) the ability to route data information into one end-to-end network solution. In addition to standard residential gateway data services, end-users are enabled with low-cost long-distance calling, faxing, and a host of advanced services including DVG-1120-to-Phone, Phone-to-DVG-1120, and DVG-1120-to-DVG-1120.

The DVG-1120 is compatible with xDSL and Cable-Modem Broadband service providers with built-in support for with DHCP Client, MAC Address Cloning, PPPoE, and multiple auto-provisioning methods.

The DVG-1120 has been tested with most major VoIP Softswitch vendors' Call Management systems. Devices such as the DVG-1120 require a Call Management system to function. These services are offered independently of D-Link, but are only required if user would like to use voice features of the DVG-1120.

For Internet connection sharing in your home the DVG-1120 offers support for up to 32 IP devices. Services offered to the internal home network are DHCP Server, Network Address Translation (NAT), and IPSEC pass-thru. Because the DVG-1120 shares your single ISP IP address, it masks the local users' IP addresses from the outside world, preventing intruders and hackers from gaining access to network resources, data files or personal information.

The DVG-1120 has voice support that includes hardware based Quality of Service (QoS), voice compression (popular voice CODECs G.711-G.729e), echo cancellation, dynamic latency (jitter) buffers, silence suppression, and comfort noise generation.

The DVG-1120 features can be remotely configured using the convenient and secure Web-based Interface. For diagnostics features, the DVG-1120 has management-at-a-glance LEDs on the front panel, and over 100 real-time statistics.

Jason Kohrs of hardwareseeker.com describes the Primus gateway like this:

VoIP, in overly simplified terms, allows one to use a standard telephone to place calls over the internet, eliminating the need for a dedicated phone line (and all of the fees/tolls charged by the phone company).

Whatis.com defines VoIP as

sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network, with more information available on their site, if desired.

by Styvas See Profile
last modified: 2007-05-28 14:56:50

Since VoIP uses the Internet and the ISP's network to reach the ATA, there may be issues now or in the future that may impact VoIP service.

Such issues may include Port Blocking or Load Balancing (data shaping). Customers need to report such problems when encountered such that appropriate action, if at all possible, can be taken by Primus.

An ISP in the USA has already been fined by the FCC for Port Blocking a VoIP provider. The CRTC would hopefully rule similarly to the FCC.

by canoe See Profile
last modified: 2005-06-16 22:37:58

NOTE: For more details, refer to:
The TBB default Emergency 911 service is tied to your Primary Number NPA-NXX (area code - exchange), and not to your billing address. Regardless of where you are physically located, 911 service will currently always connect you to an emergency call center, unless you meet the requirements for and have received Enhanced 9-1-1 service, in which case you will be routed directly to the 911 dispatcher.

Most TBB subscribers have a number in the 911 PSAP jurisdiction in which they live and have E911 service, so for the majority of subscribers this discussion is not an issue.

However, if you do not reside in the local calling area of your Primary TBB number, or if you use your TBB equipment from multiple locations, there are technical and administrative issues to resolve before VoIP 911 service can be routed and handled correctly by 911 centers.

How 911 currently works
Generally, 911 works where your call is routed to a 911 PSAP - (Public Safety Answering Point) - aka 911 dispatcher:

  • With Enhanced 9-1-1, the PSAP uses your call's ANI data to lookup a database (DB) for your address. The ANI is actually your TBB number which is always transmitted when you dial a number - it is also used for billing purposes. ANI is used because it is independent of Caller ID (CLI), which may or may not be blocked depending on your account's settings.

  • The DB address info is populated from the various Telcos - including Primus - and this is done when you get a new phone number, through the normal telephone number provisioning systems in place already (most likely through AllStream for TBB).

    FOR EXAMPLE: If you live in Toronto and have 416 TBB phone number, your 911 call will go right away to the Toronto 911 center and will transmit your CallerID. If you live in Oakville (905), and have a 416 number, your 911 call will still be forwarded to the Toronto 911 center, which may or may not be able to transfer you (or your request) to the appropriate Oakville 911 center or help you in any other way.

    The issues
    Currently, your call cannot simply be routed to another PSAP (for your street address) for the following reasons:

  • Not all PSAPS can handle or tolerate out of area code numbers that are routed to it.

  • There is no standard mechanism for PSAPs to lookup an out-of-town ANI (your TBB #). Some PSAPs search the internet etc, but this slows down PSAP service for everyone; which is a detrimental side-effect.

  • How does an Ottawa address get populated in the Ottawa 911 DB automatically - without an Ottawa telephone number ?

  • Even if Primus can route the 911 call to your local 911 PSAP, what ANI would they use? They cannot use a fake Primus ANI. Using your real ANI (TBB number) may break some PSAPs which cannot handle out-of-town ANIs.

    The solution
    With the new CRTC requirements, all TBB services are potentially nomadic (not fixed, as in Cable VoIP), so all 911 calls have to be routed to a special call center. This centre, in turn, will connect the call to the appropriate 911 PSAP.

    This means that there is 911-like service for those with phone numbers outside of their local exchange or people on the move. On a flip side, this also means a (CRTC) legislated downgrade of service to the majority of customers who have their phone number within their local exchange. Instead of going directly to their local 911 PSAP, their 911 calls first go to an intermediate call center, before being routed through - losing a few seconds of valuable time!

    Primus has received permission to have native/non-nomadic TBB users continue to connect directly to the local 911 PSAP (i.e. without the intermediate call center)- where the Primary Number is within their residential 911 calling area.

    by canoe See Profile edited by Styvas See Profile
    last modified: 2006-10-16 01:15:42

    Some Public Acronym and Glossary Links:

    • »www.voipwatch.com/dictionary.php3?op=HotDef
    • »www.answers.com/topic/voip
    • »www.vonage-forum.com/voip-acronyms.html
    • »www.voipuser.org/forum_topic_3029.html

    by canoe See Profile
    last modified: 2005-10-16 17:12:37