Jana Technology Services Blog

June 9, 2007

Understanding VOIP

Filed under: Technology — janats @ 4:32 pm

VoIP services do not use physical connections: they use protocols, with Session Initiation Protocol (SIP) the most widely accepted. At heart, it is quite dumb, merely connecting the two callers on a peer-to-peer basis and making the hardware or software handsets at either end ring, dial and connect. The full range of features open to SIP users is dependent upon the hardware at either end of the call, making this an infinitely expandable protocol. At present, it supports voice and video calling, but there is no reason why an as yet undeveloped medium shouldn’t use SIP to establish connections to other multimedia-savvy hardware at a future date.

SIP works in a similar way to email and the Web, and it goes a long way to solving the problem of interoperability between different VoIP providers. Ineen and sipgate subscribers, both of which run services using the SIP protocol, can call each other for free by appending the appropriate domain to the end of their subscriber numbers. Therefore, 2003195@ineen.com can speak to 5808085@sipgate.co.uk at no cost, regardless of the fact they have signed up with competing providers.

SIP also allows for direct connections to the outside world at little or no extra cost, using the Electronic Numbering (Enum) system. Enum should eventually unify VoIP and regular telephone networks, and as such is overseen by the International Telecommunication Union, part of the UN. It grew out of the original standard for defining international phone numbers that gave us +44 for the UK, +33 for France, and +1 for Canada and the US. This latest update, Recommendation E.164, makes little distinction between real numbers and those used by VoIP services. By the time it is fully rolled out, you will not need to know if the person you are calling has a landline or a VoIP connection, just as you do not need to know whether your contacts are BT or Telewest subscribers.

So, returning to the example above, that 5808085 sipgate number can now be dialled from a regular telephone by dropping the initial 5 and adding the appropriate UK area code, at which point it will be routed from the regular circuit-switched network to packet-switched VoIP, without the landline caller ever knowing  their outbound call has jumped ship onto the Net. Unfortunately, for the time being at least, dialling the number on its own, even from within its own local area, will not connect the call because of the way the system is configured.

The issue of numbering and how VoIP services interface with the regular telephone networks has given regulators cause for concern. In its consultation document for numbering arrangements, Ofcom outlined its belief that ‘the use of geographic numbering for [Voice over broadband] services raises a number of concerns, predominantly the impact on the available numbering resource and adequate consumer protection measures’. Its solution is the introduction of a new non-geographic subset of the 05 area code, much like the non-geographic 077 – 079 codes used for mobile phones.

Will Enum survive such intense scrutiny from regulators? Sipgate’s Steve Mancour believes so. ‘It is always dangerous to make predictions, but I think that Enum has the best chance so far. The SIP protocol is much like the hypertext protocol (http) that the Internet normally uses for web pages… designed to be scalable to huge, multinational networks, while cutting costs to a minimum. It will remain the dominant protocol for VoIP for many years.’

But SIP is by no means the only option. H.323 has been longer established and actually formed the basis of NetMeeting 2, launched in 1998 and since usurped by the largely SIP-based MSN Messenger in the years since XP first appeared. This was one of the earliest standards to define how packet-based telecommunications should work and, in particular, how Internet Protocol and ISDN should converse. It gained widespread acceptance, and H.323-compliant equipment is still available, representing SIP’s only realistic high-end, non-proprietary competitor.

However, passing voice traffic over a network is a far from trivial feat. It is now well known that the Internet works by splitting data into separate packets, which are then sent across the network using whichever route is the most efficient. This route can change by the second, so there is no guarantee each packet will immediately follow the one that went before it. Once they reach their destination, they must therefore be put back together in the right order and translated into a form we can use. Doing this with a constant two-way voice stream is a highly complex operation, and can result in slight latency.

VoIP protocols employ a number of techniques to minimise this effect, the most effective of which is to transmit just the audible part of each conversation. For much of any call, at least one of the participants will be silent, and so only the incoming stream will be transmitted. A poorly configured microphone without noise cancellation, however, will continue to transmit background noise even when a participant is not talking, making the data exchange more complex.

SIP in the Wireless World:

In the wireless world, SIP has not been very popular so far due to a number of wireless network limitations. GPRS and other first generation wireless IP packet networks are too slow and the latency of the connection was too high. In addition, speech algorithms used by current SIP implementations use inefficient codecs which require a substantial amount of bandwidth. 3G networks such as UMTS offer higher bandwidths compared to earlier networks and are thus able to carry SIP voice calls over the air interface. A SIP call, however, uses around five times more bandwidth then a traditional circuit switched mobile voice call for which very bandwidth efficient codecs are used in the radio network. This fact together with the openness of SIP for the user to choose the operator of the SIP server himself explains the reluctance of wireless operators to support the application of SIP services in their 3G networks.

The Future of SIP in the Wireless World:

In the near future, SIP clients will mostly be adopted on GSM/UMTS/Wifi smart phones such as the Nokia N80, where they can be used to make phone calls over a Wireless LAN access point connected to DSL or a company network. My current Smartphone, the Nokia E61, already has this functionality built in.  In effect, a SIP client in the mobile phone can replace the fixed line phone at home and I am actually waiting for the day when I can use a single phone at home and when underway – I actually can if I want Bt to be my provider.

When leaving the office or home, a SIP client can still be used for voice calls but many operators (carriers) try to restrict SIP for the reasons discussed above. Very near term evolutions of 3G networks to technologies like HSDPA (High Speed Data Packet Access) might change these policies in the mid term.

IDC said 124.3 million homes worldwide had broadband access at the end of 2004. This was up 47 per cent on the year before and opened up a vast market for Internet telephony. The potential for global point-to-point PC calling is growing on a daily basis, giving the nomadic VoIP user more opportunities to hook up their personal number wherever they stop for the night.  This is not complimented, by VOIP providers such as Skype, with the ability to route the call through to another number if the VOIP number is offline.

VoIP is unlikely to totally replace conventional setups any time soon. Currently there are just too many problems inherent in using an entirely packet-based medium for voice. Beyond the issue of emergency calls, there is availability to consider. The public phone network does not power VoIP hardware the way it does with conventional handsets, so will be out of action if your power blacks out, which could be when you need it most.

This however misses the point.  Most people are using VOIP to supplement their traditional call services, not replace them.  Alsoost VoIP services also cannot host a fax machine which is an important consideration and potential opportunity for us to take advantage of at Yac.

Each of these factors counts against VoIP, which will remain a complementary service for most users, with corporates retaining at least a handful of conventional lines as entry points to the legacy network for fax machines and monitored alarms. Home users too will be given the choice of reliable, conventional calling and cost-saving, innovative VoIP. Even the providers accept this, with Skype’s Sarah Myers explaining that: ‘[in the same way] email is free but people still pay for and use the fax, we feel that voice calls using Skype could be a free alternative to other telephony for anyone using the Internet.’

Assessing the VOIP Implementations

Perhaps the best-known service, and certainly one of the easiest to set up, is Skype. With SIP-based services, you may find yourself filling in long configuration forms, unless you are using a pre-configured router. But by using its own proprietary format, Skype is able to provide a fuss-free installation. The settings are built into the software itself, so all you need do is pick a username and hook up your speakers and mic. It is as simple to complete as signing up for MSN Messenger.

Call quality can be variable, but usually rivals a traditional circuit-switched connection and, with a wide range of software plug-ins and hardware peripherals reaching the market, it is a flexible option that will certainly grow over time, especially now that it has the  muscle of Ebay behind it.

The best conventional international deal seems to be  provided by Pipemedia. Its PipeCall service incurs both an installation fee and a monthly charge of £2.99, but once paid the European, American and Australian call costs are unbeatable. At less than 1p a minute for some European numbers, and with a dedicated incoming number thrown in (which allows mobile/fixed routing), it seems to be  a no-brainer for heavy users.

However if mainly calling within the UK. The £2.99 a month plus tax and the installation fee means the payment will be about £60 in your first year.   This gives us a good pricing benchmark for a VOIP service that also provides inbound call services. 

You could easily undercut this cost  with sipgate. For the same price as a subscription to PipeCall, you could make 5,019 minutes of UK calls using sipgate’s tariff-free service. You will have to provide your own headset or handset, but you do get a dedicated number with a UK area code and free voicemail thrown in to boot, but no number routing for free,so inbound call management is poor.

The  sipgate1000 tariff is that for  £5.90 a month, your annual costs will be £70.80, but will give you 12,000 minutes   of outbound calls  to UK landline numbers, which is more than enough for anyone running a business from home, and almost half the price of BT’s annual subscription to Broadband Voice Anytime.

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  1. I found the part about assessing the VoIP implementation to be most interesting. In the cost you mentioned that from Pipemedia for international calls that it can be lees than 1p to some number. That is dirt cheap! I don’t see how anyone would want to pass that up.

    Comment by VoIP Reviews — June 13, 2007 @ 5:05 am | Reply

  2. […] don’t have to be a rocket scientist to understand VOIP, but it might help. Voice over Internet Protocol is used by millions through out the world and many […]

    Pingback by VoIP News » VOIP for Beginners — June 13, 2007 @ 5:54 pm | Reply

  3. Nice introdution to some VoIP topics here. Thanks for the post.

    Comment by parkpost — August 17, 2009 @ 11:35 am | Reply

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