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» What is VOIP?
What is VoIP
Voice over IP, or VoIP for short, is a method of delivering video and voice over a data network like the Internet.
VoIP converts the voice signal from your telephone into a digital signal that travels over the Internet, and then converts it back at the other end so you can speak to anyone with a regular phone number. When placing a VoIP call using a phone with a VoIP adapter, you will hear a dial tone and dial just as you always have.
Because VoIP is digital, it may offer features and services that are not available with a traditional phone. If you have a broadband Internet connection, you need not maintain and pay the additional cost for a line just to make telephone calls.
With many VoIP plans, you can talk for as long as you want with any person in the world (the requirement is that the other person has an Internet connection). You can also talk with many people at the same time without any additional cost. (top)
How does VoIP work?
The internet was originally designed to send small packets of information (data) from one computer to another. What VoIP does is it takes voice from one point, breaks this down into a lot of small packets, transmits these via your internet connection and puts them all back together again so the information can be heard at the other end. Thus allowing a telephone conversation to take place. (top)
How is this different to my normal telephone?
The basic principle of the PSTN (Public Switched Telephone Network) network is that two users are connected to each other by a length of cable. The conversation is transmitted in one stream down the cable. When you make a call with VoIP the conversation is broken up into packets of information which may travel via different routes to reach their final destination. The packets are then put back in order when they reach their destination allowing sound to be heard. (top)
Is the quality on VoIP as good as a normal telephone?
Under normal circumstances the quality of a conversation using VoIP will be indistinguishable from a conventional telephone service. However, from time to time the internet may slow due to large volumes of traffic. At these times you may experience a connection more like a mobile telephone conversation. Why not test your connection now: Click Here (top)
Can I call someone who doesn’t have VoIP?
You certainly can. Most service providers offer the opportunity to add credit to your VoIP account. This is needed when calling a non VoIP service. Call cost vary so shop around or ask us. To make a call you simply dial the number including area codes and you will be connected. (top)
Can someone who doesn’t have VoIP call me?
Yes. If you have subscribed to a VoIP provider who supplies a direct inward dialing (DID) number you will be assigned a local number just as if you had a normal phone. (top)
What do I need to make a call using VoIP?
You will need one of three things to enable you to use VoIP.
1. You can download a Softphone and use your existing speakers and microphone to make the call, although the call will be enhanced by using one of our telephone handsets Link.
2. You could also purchase an IP Phone which resembles a normal phone or you can purchase an ATA which allows you to plug in any normal phone and begin making calls.
3. You will need to subscribe to a service provider who will make the connections for you (similar to a normal Telco). (top)
What’s the easiest and cheapest way for me trial VoIP?
If all you want to do is speak with a friend or family member for free we recommend you both download the firefly software. This software will even work with dial up connections! Although if using dial up, be aware that voice quality will not be as good as it could be. We have no association with firefly and make this suggestion in the interest of saving you money. If you find this service benefits you why not revisit us and buy a telephone handset to enhance your call quality. (top)
Can I connect to VoIP using a Dial up connection?
VoIP can be used with a dial up connection, however it is not recommended if you are looking to compare the service with your normal telephone connection. If you want to speak to someone for free this is definitely an option but it would be unfair to suggest the quality you would experience using a dial up connection is anything like using dedicated IP phones or ATA’s with a broadband connection. Many people still connect using a dial up connection. If you would like to do this we suggest you consider purchasing one of our telephone handsets which will greatly improve the quality of your calls. (top)
I am thinking of getting a USB phone, how do they compare?
USB phones plug into the USB port of your computer. I have read some very good reports about them; however I also believe they sometimes have compatibility problems with various software. That is if they are not purchased from the service provider for whom they will be tailored for. The advantage an ATA or IP phone over USB phones is there is no need for the computer to be turned on for the phone to work. With a USB phone voice traffic is processed by your computer before being transmitted or received. With an ATA or IP phone all the processing is done by the phone or the ATA. (top)
Can I keep my current phone number?
Not yet, however some service providers are working on this and expect to be able to offer this feature in the very near future. (top)
Can I use the VoIP line for my Fax?
Some providers do offer this feature however the service cannot be relied upon at this stage. This technology is advancing at a rapid pace so it most likely won’t be long before we see huge improvements. (top)
Do I need a broadband connection?
It is recommended you have a broadband connection preferably connected to your PC via an Ethernet port not by USB. (top)
How much does it cost to subscribe to a VoIP provider?
Most providers offer a no fee service where you will only pay for the calls you make. These providers will either require some credit card details or the alternative is to use a pre-paid service where you add credit to your account for your use. If you choose to only speak with others on the same service your call costs will be free. (top)
How much bandwidth does Voip use
The codec you decide to use will determine how much bandwidth you are likely to use. As a guide the smaller codec's use about 30MB per hour while the larger ones use up to 100MB per hour. (top)
What is quality of Service (QoS)
QoS, VoIP and Home Broadband
For the purposes of this forum, QoS is used to minimise the impact of other traffic on VoIP traffic. Much of the forum discussion concerns VoIP over Home Broadband connections and this summary is written in that context.
QoS works best when you have complete control of the network between the communicating endpoints. However, the Internet comprises many entities, all providing "best effort" delivery, so this control is not something a Home Broadband user has. In this situation, QoS can’t guarantee VoIP performance, but it can help minimise problems (at least on the ISP link).
Network Impact on VoIP
According to a Cisco White Paper, the key network issues that VoIP traffic is sensitive to, are:
Latency
# Latency is the average travel time it takes for a packet to reach its destination.
# If there is too much traffic on the line, or if a voice packet gets stuck behind a large data packet (such as an email attachment), the voice packet will be delayed to the point that the quality of the call is compromised.
# The maximum amount of latency that a voice call can tolerate one way is 150 milliseconds (100 milliseconds is optimum).
Jitter
# In order for voice to be intelligible, consecutive voice packets must arrive at regular intervals.
# Jitter describes the degree of variability in packet arrivals, which can be caused by bursts of data traffic or just too much traffic on the line.
# Voice packets can tolerate about only about 75 milliseconds (40 milliseconds is optimum) of jitter delay.
Packet loss
# Packet loss is a common occurrence in data networks, but computers and applications are designed to simply request a retransmission of lost packets.
# Dropped voice packets, on the other hand, are discarded, and not retransmitted.
# Voice traffic can tolerate less than a 3 percent loss of packets (1% is optimum) before callers experience disconcerting gaps in conversation.
Minimising Network Impact
QoS can help address these issues and minimise their impact. On consumer level routers, Prioritisation and/or Traffic Shaping are the commonly available QoS tools.
Prioritisation
# Packets are marked with a priority based on characteristics such as IP/MAC Address, Physical Port, and/or TCP/UDP Ports.
# VoIP devices (ATA, IP Phone, etc) can also mark packets, although it depends on the Router’s capability to recognise and act on it.
# High Priority traffic is favoured on transmission.
# Does help with upload traffic on the link to your ISP.
# Does not help within your ISPs network and beyond as they will either ignore and/or strip/overwrite these user set priorities.
# Does not help with download traffic as your ISP will deliver traffic to you according to their own policy - probably “as it arrives” with no prioritisation.
# You cannot affect the priority of download traffic over your ISP link by setting priority on your Router.
# The points above don’t mean that an ISP isn’t implementing some kind of QoS that may happen to benefit your VoIP traffic. It’s just pointing out that it has nothing to do with your configuration and is outside your control.
# ISP provided VoIP is an exception: they will mark packets and priorities them within their own network and on your download link.
Traffic Shaping
# Traffic Shaping manages bandwidth to ensure enough is available for VoIP traffic.
# Can help manage download traffic (and upload traffic if desired – useful if Prioritisation doesn't do enough).
# Traffic Shaping usually drops packets for high traffic flows based on a configured bandwidth profile.
# Works best with applications using TCP protocols because of the built-in backoff mechanism that will slow down traffic if packets are lost. Applications using UDP have to implement their own mechanism to handle lost packets and may not slow down at all. For example, VoIP RTP streams don't slow down with lost packets.
# Many options for implementation including your router or other traffic shaping device, software on each PC, or built-in shaping available in some download, P2P, etc clients.
Other
# A good ISP will help: If your ISP has overloaded links or other issues that affect latency, jitter or packet loss, your VoIP traffic will suffer. This applies for each and every ISP between you and your VoIP provider.
# Higher upload/download speeds will help: If you have more bandwidth, your uploads/downloads will have less effect on VoIP traffic, whether you have QoS or not. Your ISP link is probably the slowest in the chain and will benefit most from Bandwidth/QoS improvements.
# A Simple solution: Don’t run anything that uses bandwidth when you're making a VoIP call. :)
VoIP Protocols
Prioritisation and shaping should be done with knowledge of the protocols being used on your connection, both for VoIP and for other applications. Most devices discussed on these forums are SIP or IAX based. As an example, here are the protocols used by SIP based VoIP:
SIP - Session Initiation Protocol.
# Signalling, including Call Setup, Termination, etc.
# UDP Port 5060 (most common) - technically any port and/or TCP could be used; check the configuration, especially if your device can handle multiple SIP providers).
RTP - Real-time Transport Protocol.
# Carries the data (audio media stream).
# UDP Port x (port depends on the VoIP device; check the configuration; a range is usually configured).
RTCP - RTP Control Protocol.
# Carries information about the RTP media stream, including quality.
# UDP Port x+1 (next port after that used by RTP; depends on the VoIP device; determined by the RTP configuration). (top)
With Asterisk as your telephony switching platform, you'll not only have a high-class PBX replacement. Asterisk is much more than the standard PBX. With Asterisk in your network, you can do telephony in new ways.
- Connecting employees working from home to the office PBX over broadband connections.
- Connecting offices in various states over VoIP, Internet or a private IP network.
- Giving all employees voicemail, integrated with the Web and their E-mail.
- Building interactive voice applications, that connect to your ordering system or other in-house applications.
- Giving access to the company PBX for business travellers, connecting over VPN from airport or hotel WLAN hotspots.
The Ability to Interface with Normal Telephone Lines
Asterisk is an open source software PBX, created by Digium Inc. and a continuously growing user and developer base. Digium invests in both developing the Asterisk source code and low cost telephony hardware that works with Asterisk. Asterisk runs on Linux and other UNIX platforms with or without hardware that connects your server to the traditional global telephony network.
The two major advantages the Asterisk approach are:
1. Significantly lower costs (since the software is free).
2. Rapid development: today thousands of people all over the globe work with Asterisk, many of them contribute to the code. Asterisk literally evolves and improves from day to day.
As a result, by using Asterisk it is possible to build high–end telephony systems for a fraction of the cost incurred when building them in the traditional way.
Asterisk is fully capable of working with IP telephony as well as with POTS (Plain Old Telephony Service) and analog telephones. This new approach to the world of telephony will change the rapidly–expanding iPBX market dramatically in the near future.
Asterisk Features
Asterisk is feature–rich and is growing rapidly. Apart from basic capabilities, such as call routing (including DID – direct inbound dialing), call forwarding, music on hold etc., Asterisk can also serve as a conference bridge, send voice mail to email, serve as an IVR and much more.
Asterisk Usage
Many different types of users, from private and small business implementations to large call centers and service providers, use Asterisk today worldwide. Since Asterisk is open source, it can be implemented as a PBX, or be used for a single purpose, such as voice mail or a conference bridge for an existing telephony system.
Asterisk as a Traditional PBX
Asterisk can be used as a PBX for traditional analog telephony. This means you can upgrade an old telephony system without the high cost of changing all of your telephone sets to IP phones. You can enjoy all the features without investing in additional equipment. Asterisk also allows you to gradually start using IP service providers and IP telephone sets in conjunction with the old equipment. In order to use Asterisk as a PBX for traditional telephony, it is necessary to use specific hardware with telephony interfaces, such as channel banks, PCI cards, or small gateways.
The Future of Asterisk
Asterisk is growing at an extraordinary rate. VoIP guru Jeff Pulver states: "They are developing a sophisticated PBX on a PC with the (capability) of a $100,000 PBX…It will be a world class PBX that runs on Linux. You can have a PBX for the cost of a PC. Jon ‘Maddog’ Hall, president of Linux International, states: “I predict that over next three years, VoIP using an open-source solution, such as Asterisk; will generate more business than the entire Linux marketplace today." interfaces (top)
1. Choose a PC that will satisfy your requirements
a) The basic PC offers up to 50 IP extensions and VoIP
It allows for one board to be used for either:
i) Analogue extension e.g. portable phones, fax etc
ii) A PSTN board or
ii) A PSTN board or
It can also be used as a stand alone Voicemail with Auto Attendant
Do you require RAD as a backup?
What about extra RAM should your system need more memory.
Select the extra features that are not standard that will add to your company professionalism.
b) A larger PC that allows for extra cards so that you can have PSTN, ISDN (Multiline), and Analogue extensions.
Do you prefer a standalone or a Rack Mount?
2. How do you want to communicate?
Handsets or Softphones
Select the IP Phones that suit your budget
Do you require IP Headsets or adaptors to add to your existing headsets?
3. Select your Network card
Choose the Network cards required:
PSTN
ISDN
Analogue
Do you need an Ethernet Router- this is to take the IP handsets
What about a UPS in case of power failure
4. Are you receiving the best deals on your communication costs?
Send in your bills for IM I.T. to reduce your charges
Do you want to keep some PSTN lines- reduce your monthly line rental and charges- we would recommend ISDN (Multiline)
Save on the line rentals by switching to VoIP.
Let IM I.T. Assist with a reliable ISP (Internet Service Provider)
What about your mobile communications reduce calls to mobiles, Blackberries, PDAs, remote users.
5. Do you want to install yourself or let IM I.T., we have the experience to do it quicker
Do you want to take maintenance?
Once you have filled this in and checked your monthly payments – email to IMI.T.
Asterisk is an open sourced software that can be down loaded by anyone from the web. Like most software the basic information is provided however many man hours are required to convert the software into your telecom requirements. IM I.T. through its developers have allocated thousands of hours into R & D to maximize and improve performance.
Features as Standard.
IM I.T. can add the IP PBX system to your current PABX for you to try before totally converting. You can retain all your existing features and add the IP PBX through your PABX and utilize its extra programming. Ring our Help Desk for a quote – try configuring yourself – then contact us and we will talk you through it.
Every user can have their own voicemail, unlimited calls not limited by the number of ports as on a standard PABX, which usually comes with a limited 2, 4, 8 or 16 ports. Prices nearly double with each increase in size - there is no comparison to the value offered by Asterisks.
Voicemail ensures you are contacted around the clock; it is an all in one call management system.
Your voicemail system provides effective and affordable call management for your business 24 hours a day, every day of the year.
Your company and staff will never miss a call again. Provide limitless on-dial options- all of them relevant to the extension you called. Transfer to a PA or associate- transfer to a group of phones – transfer to a mobile or a branch office.
There are many types of claims on Unified Messaging however true Unified Messaging is Voice, emails and faxes all on the one screen generally your Microsoft Outlook, Asterisk provides all these features if requested for an investment far less expensive than third party products covered by tradition POTS (Plain old telephone systems).
Despite the way email has overtaken fax, sales of fax machines continue to grow. – Why have a manual fax machine when you can receive your faxes on-line.
Asterisk delivers your faxes directly to your inbox in Outlook or Notes- no one has the opportunity to read your faxes without your permission.
You are notified immediately by the inbox and the phone message light.
With Asterisk it all happens in the inbox. Your voice and fax messages are in there with your emails. Click a voice message to have voice mail ring you and play the specific message you clicked. From there you manage them just as you do with email. Archive, print, forward, reply or delete them.
You can be notified by email of the successful transmission of a fax – or a failure, along with the reason.
Whether you require screen pops, to your Access database or other Microsoft integration numerous sites have had their programmers integrated by the IM I.T. developers and installing Engineers. This is an extra cost depending on your requirements and our Help Desk is able to give firm quotes to satisfy your requirements.
Many companies have their own arrangements with Hardware distributors or manufacturers and have policies to stay with this supplier. IM.IT. is happy to work with your requirements; however there is a handling cost and also the configuration of the system software is added to the system. It is a requirement that the PC supplied must be new and covered by a manufacturers warrant. If you supply your own PC there is a margin added to the PC to allow a margin for IM.I.T, to provide your service.
PSTN has been around for a long time and prior to its use for ADSL and other users did have a used by date and compared to other services such as ISDN Multiline and VoIP has a limited life. Especially as it can be seen by Telstra’s constant news reports of lost revenue.
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