Frequently Asked Questions

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1. What is Voice over IP?

Voice over IP or VoIP is a set of standards, CODECs, protocols and hardware devices that enable people to transfer telephone calls and voice data over a network or the Internet. The major advantage of voice over IP is the ability to place calls over a data network providing huge cost savings on hardware and implementation, interoffice and international calling charges, and long-term support costs.

For additional information on VoIP and VoIP technologies please use the following links:

VoIP Solutions by Bitware Technologies

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2. What is Asterisk PBX?

Asterisk is an open source software private branch exchange (PBX) used to route telephone calls through a data network or the Internet. Additionally an Asterisk PBX connected to the public switched telephone network (PSTN) provides gateway services connecting IP phones on an internal voice network to the public telephone system.

Asterisk provides a full feature set including voice mail, automated attendant, interactive voice response (IVR), conference calling, and automated call distribution that are typically only found on expensive traditional PBX phone systems. Asterisk supports a host of VoIP protocols including IAX, SIP, and H.323.

For additional information on Asterisk please use the following links:

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3. How can BitWare Technologies save me money with Voice over IP?

Cost savings begin at implementation with a Voice over IP solution from BitWare technologies. The hardware and support requirements of an Asterisk based PBX solution deployed by BitWare Technologies are much less that of traditional proprietary PBX solutions. We can save you tens of thousands of dollars on a small to medium deployment and even more on larger deployments over traditional solutions.

Leverage your existing data infrastructure and send your voice traffic at a compressed rate over your local network. A BitWare Technologies Voice over IP Solution can connect multiple sites and endless users to one system--or multiple systems with the ability to share dial plans--eliminating toll charges for interoffice dialing and international dialing across the same private network. Furthermore, the ability to compress voice signals and use existing infrastructure means that our customers can support more calls at a lower cost with less infrastructure than a traditional system.

We continue to save our customers money with support solutions that have value and responsiveness in mind at a reasonable cost. We
challenge all of our customers to shop around for a traditional system that can provide the full featured cost saving value of Voice over IP phone system from BitWare Technologies.

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4. What are some reasons to choose a Voice over IP solution from BitWare Technologies?

Top 10 Reasons to Choose a Voice over IP Solution from BitWare Technologies

  • Save money on implementation and continuing cost of ownership.
  • Reduce or eliminate overseas dialing charges to your international locations.
  • Eliminate interoffice dialing costs.
  • Leverage your existing data infrastructure to connect multiple sites and users to one phone system.
  • Implement leading edge technology to the delight of your employees and customers generating greater productivity, saving you money and generating additional revenues.
  • Receive a full featured PBX solution at a reasonable cost.
  • Build a multimedia data solution that handles all of your voice, fax, and video conferencing needs.
  • Work with experienced IT professionals, versed in data networking and voice solutions that are dedicated to making your technology vision a reality.
  • Integrate custom software solutions into your new Asterisk based VoIP phone system, further enhancing and personalizing the system to your business needs.
  • Cut the strings to your old phone system provider and costly support fees, freeing revenues for your business.

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    5. What is the rate of audio compression used with VoIP?

    There are several methods for audio compression used with VoIP. Some of the most commonly used are G.729, G.711, G.723.1. G.729 is the most prevalent standard of voice compression used with VoIP and typically compresses a 64 kbit/s voice signal to 8 kbit/s in 10 millisecond chunks, however, extensions exist to compress data to 6.4 kbit/s and 11.8 kbit/s with marginally better or worse voice quality respectively. In situations where G.729 is not the preferred method of transport such as for fax tones, G.711 can be used to transmit data with a 64 kbit/s stream sampled at 8000 times per second. Where compression is desired over quality G.723.1 can be used at 6.3 kbit/s and 5.3 kbit/s in 30 millisecond chunks.

    For more information on audio compression standards please use the following links:

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    6. What is SIP?

    The Session Initiation Protocol (SIP) is currently the leading signaling protocol used with Voice over IP. SIP provides call processing features currently present in the public switched telephone network (PSTN). SIP provides SIP enabled end devices with the familiar dial tone, call dialing, busy signal, and tones associated with a traditional phone system. Additionally, SIP allows for peer-to-peer connectivity allowing for a simplified network core and increased scalability.

    For additional information on SIP please use the following links:

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    7. What is IAX?

    The Inter-Asterisk eXchange (IAX) protocol used by the Asterisk PBX to route voice data over IP and provide in-band data and signaling that is used for voice connections between Asterisk PBXs and IAX enabled devices. IAX can be used to route almost any type of data providing great advantages when used with IP based video conferencing devices.

    For additional information on IAX please use the following links:

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    8. Can I use Asterisk with traditional analogue phones?

    Yes. Asterisk can scale to support almost any voice platform including analogue phones. In situations where a switched Ethernet data network is not feasible €”such as a hotel without a data network €”Asterisk can provide service to analogue phones with the addition of analogue channel banks to the system. This provides customers with an existing analogue phone system and phones a low cost option for migrating to an Asterisk based PBX.

    For additional information please use the following links:

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     9. How many phones can be supported by an Asterisk based VoIP phone system deployed by BitWare Technologies?

    BitWare Technologies deploys voice solutions that can scale to the needs of any enterprise. Our smallest system supports up to 100 phones. For larger deployments our solutions are custom architected and scaled to meet the needs of the business.

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