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VoIP by vivi07

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									VOIP
Voice Over Internet Protocol (VOIP) or Internet Telephony has been around for over 10 years through systems like Microsoft's NetMeeting. The availability of faster connections has made it more viable and the standards have improved to give better quality and compatibility. In some countries, for example China, VoIP offers low cost telephony to the masses. In others it is a way to make free calls to VoIP-savvy friends, to provide an extra phone line or to replace the analogue landline service. This article gives an overview of VoIP from a UK perspective, discuss it or ask questions in the VoIP section of the forum.

What are my options ?
VoIP calls can be made between two PCs equipped with softphones like the X-lite or the proprietary PC-only service from Skype - in both cases you use a headset, mic/speakers or USB "phone" to talk and the PC does the translation of voice audio into VoIP packets. If you prefer to use a standard telephone, or convert your house over to VoIP, then an Analogue Telephone Adapter or ATA may be a better choice. These have an ethernet socket to connect to your network and a phone socket to plug in the phones. A popular combination is to plug a DECT cordless phone into an ATA. Some ATAs include a connection to a landline phone, others can handle more than one account and/or more than one phone or network circuit, the Sipura range offers several options. If you need to connect several phones to a VoIP system and provide several virtual "lines", or if you want to use different VoIP services for different tasks than a software switchboard like an Asterisk server or SIPswitch may be necessary. These can do music on hold, least cost routing, voicemail, internal calls and the usual switchboard facilities used in an office. You still need a softphone or ATA/phone combination to use an Asterisk server, as the latter is literally just a computer sat on a network. Asterisk servers can be equipped with so-called FXS phone ports and FXO line cards to connect to extension phones and exchange lines. In summary, the major options are software-only, ATAs and switchboards.

Which one is for me ?
If you are happy to make and receive all your VoIP calls at your PC then a software solution will work for you. Skype and softphones like X-lite are available for pocket PCs and laptops as well as desktops. You may be able to extend the audio range with a bluetooth headset connected to the PC.

If you want to use a normal phone but have the calls go over VoIP then you need an ATA. You could also invest in a VoIP phone, which is a telephone that plugs directly into Ethernet and contains the codec circuitry. If you are an enthusiastic VoIP amateur or if you want to run an office or number of offices on VoIP then an Asterisk server is the way to go. A CAN could perhaps provide Asterisk based services to its customers and trunk the calls to a VoIP provider to make best use of backhaul bandwidth.

How do I make free calls ?
In general you can only make free calls to other VoIP users, although some services like Voipcheap that are starting up are providing free calls to landlines initially. Skype is a system based on free peer-2-peer calling using usernames. Most VoIP services provide free calls between users of the same VoIP service, for example FWD , and many of these provide gateways or peering to other services to allow free calls to users of other services. Skype uses its own proprietary systems and doesn't integrate with anything else. SIP based services allow calling between IP addresses or by using the SIP address which is like an email address - for example 123456@fwd.com - which is translated into an IP address by a SIP server. This allows you to plug in an ATA or VoIP phone anywhere and receive your calls, for example a US office could receive calls to a UK number by having a UK SIP account.

How do I make and receive calls to normal phones ?
Normal landline (PSTN) phones can make calls to VoIP phones providing the VoIP service provider gives you an incoming landline number. This might be a nongeographic number with a prefix like 0870, 0844, 0871 or the VoIP specific 055/056 ranges or a geographic code like 020 for a London number or your local 01xxx code. You may even want to have a US or Hong Kong number, to allow friends or customers there to make cheap calls to you. Making calls out to landline phones again depends on the service provider concerned giving you the access and providing a billing or pre-payment system to fund the calls. Such calls are not necessarily cheaper than the equivalent indirect or CPS call rates available over a BT line in the UK. Some providers, for example 18866, provide outbound VoIP at good call rates but don't do incoming. To get the absolute best deal you may need equipment that can make calls on one provider and receive on another, or allow you to select which provider you make cals through depending on the type of call. SkypeOut and Skype In are add-ons to the basic Skype service to allow you to make & receive calls from landline phones.

Do I need a service provider ?
If you want someone else to route your calls onto the landline PSTN network, or allow you to receive PSTN calls, or to give you a SIP address or provide P2P VoIP calls then yes you need a service provider. If you just want to call a friend in the states using each others IP addresses then you don't. You may also want to become a service provider and run your own server and landline interconnects. The market is fast evolving so this isn't the place for listing & comparing service providers. To illustrate the range on offer, at the time of writing SIPgate provide geographic incoming numbers in the UK so your neighbours can call you at local rates. Vonage and BT BBV provide subscription services with bundled calls and ATA hardware. These are pre-configured (or self configuring) and locked to the service, so there is a downside but it is an easy route to a functioning service. VoIPtalk offer both SIP and IAX based accounts and a range of add-on services including 0870 and 0845 incoming numbers and fax to email gateways.

What about the BT line rental ?
VoIP users connected by cable internet, wireless broadband, satellite or anything other than ADSL have the option to dispense with their BT land line and stick to VoIP alone. It is worth thinking through the reliability and safety aspects - for example does your VoIP service work with the power off and does it allow 999 calls - but the option to get rid of the landline or have only one is there. ADSL users will still need to pay line rental for the line, and its worth saying that UK calls in the evening/weekend and to some types of number like 0870 may actually be cheaper via the landline.

Which protocol ?
SIP is a standards based protocol used by most "heavy duty" VoIP providers. IAX is also an open standard protocol that evolved out of the Asterisk server project, it is simpler to handle through firewalls. IAX softphones and ATAs are available but are far less common than SIP versions. Skype is proprietary and restricts you to communicating with other Skype users.

Is it easy to set up ?
In general VoIP is no harder to set up than a broadband connection. The hard parts are usually where a firewall has to be adjusted to allow the VoIP traffic through, or "port forwarding" has to be configured to get an incoming call routed to the right internal IP address. These issues are common to most applications that live on NAT local IP addresses or behind firewalls, and shouldn't put people off. Systems like STUN are evolving that work around these issues and I have personally found that recent services have "just worked" when plugged into a firewalled and NATted LAN. In a way the hard part is finding a service with the right blend of facilities, call charges and incoming numbers that is compatible with what your VoIP friends use. A compromise may be to use Skype for PC based person to person free calls and an ATA for calls to/from landline phones.

Finally, a word about Bandwidth.
VoIP involves digitising the voice signal into a data stream, packeting it up into IP packets and routing it to the other caller, where it is unwrapped and turned back into an analogue voice signal. The analogue to digital parts use "codecs" or coder/decoders which use different amounts of bandwidth to carry calls with different qualities. GSM mobile phones use a 14.4 kbits/s channel for voice, BT landlines use 64 kbits/s and a typical VoIP call using a G.711a codec will use about 100 kbits/s or 40% of the upstream rate of a typical ADSL line. VoIP uses more bandwidth than a standard call because there are extra overheads of packeting it up to go over the internet, rather than streaming it down a dedicated circuit. By reducing call quality and using codecs with more compression the amount of bandwidth consumed is reduced. Asterisk servers can establish a trunk connection to a VoIP service provider which also reduces the total bandwidth while maintaining quality. A VoIP call using high quality codes will use 100 kbits/s of bandwidth each way, both up and down. If two VoIP calls are to run on an ADSL upstream of 256 kbits/s then some form of Quality of Service (QoS) priority will need to be given to VoIP otherwise other activities on the network may disrupt the speech. ATAs and VoIP routers with QoS features are becoming widely available. So there is a trade-off between bandwidth and quality, and you also need to ensure that the codecs used are compatible. Asterisk servers will do codec translation but an IP to IP direct call will need to have compatible options at each end. Article by Phil Thompson 15/09/05 from Wray Village Parish Plan Last Updated (Thursday, 15 September 2005)


								
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