WhichVoIP Blog

Just a quick note folks regarding Lingo setup, in particular for the Linksys SPA-2102. 

I use Lingo for home phone service (have done for years) but recently had to get my VoIP box updated and received a new one in the mail free of charge from Lingo and this one was from Linksys (SPA2102).

The recommended setup in the users guide is as follows:

(Cable/DSL Modem) –> (VoIP Box) –> (Router if needed) –> (PCs)

as this allows the VoIP box to provide QoS (i.e. it can bandwidth limit the data from the router to ensure your voice is top priority when it comes to your broadband bandwidth).

However in the QUICK guide it shows the recommended setup as follows:

(Cable/DSL Modem) –> (Router) –> (VoIP ATA) –> and PCs hanging off router too if needed

This does NOT provide QoS so be careful if you are making calls at the same time as the internet is being used!!

One other note by the way (this took an hour or so to figure out so may save you some time)

The new Lingo VoIP box they are giving out is the Linksys SPA-2102. It is a VoIP ATA and router combo and is a decent piece of hardware.

HOWEVER, it lacks one thing that may be important for you folks that work from home. It does not allow VPN through the router! This is not ideal but there is a fix I stumbled across. Enable DMZ (De-militarized Zone). This essentially disables the router function in the SPA2102. This is fine for me as I have a router connected to the SPA-2102 that DOES allow VPN pass-through. This works for me but may not be good for all situations so up to you

Cheers.
Andy

P.S Reason I needed new box if you are interested. For some reason my old VoIP box would not allow me to join conference calls, I could phone in but it would not accept the passcode to enter the meeting.

Fed up paying a fortune for your business phone service?  May be worth taking a look at a business hosted VoIP service: 

Here is a summary of the important benefits and advantages of a hosted VoIP model. Think of these not just at face value, but how they are additive to what VoIP brings on its own. It is the combined set of benefits that leads us to conclude that hosted VoIP is an ideal solution for micro businesses looking to move beyond conventional telephony.

Benefits of Hosted VoIP
Only a one-time expense for IP-enabled telephones to support VoIP service and features. No capital expense is required and you are not locked into an asset with a long depreciation cycle.

Ideal For Multiple Locations
Hosted VoIP is very flexible for supporting branch offices, remote workers, and new sites as the business expands. All that’s needed is to add handsets as you go when setting up a new office or adding more locations to your network.

Business Continuity
Hosted supports the 24/7, mobile, on the road realities of today’s business world. Business continuity is a core element of hosted VoIP, as the network services are housed in rock solid data centers. This also includes disaster recovery, a capability that no premises–based system can provide. In the event of a natural disaster or malicious network attacks, your messages and directories are never lost, and the business can carry on under all conditions.

Reliable Service
Regardless of how busy your phone lines get, hosted VoIP always works. Your customers and employees will always be able to get dial tone, and they’ll never get a busy signal, even during peak periods.

Future Proof
VoIP is still a new technology, and we are just beginning to see what is possible. With hosted VoIP, you will always have the latest features and services, and will never need to worry about implementing them with your telephone system. For the first time, micro businesses have a solution not only to match what bigger competitors can do today, but to keep pace as technology advances.

If this is of interest to you for your business phone service, submit a request for a free quote NOW!

Andy

Came across a good blog today from Tom at TMCNet. 

There are a couple of good reasons for moving to T1 for your broadband data and voice needs.  The first of course is the Quality of Service (QoS) you get from T1 service providers.  The second is symmetrical uplink and downlink speeds of 1.544Mb/s.

Tom recently moved to T1 and had some fun getting full bandwidth on his uplink.  Turned out his Linksys router had QoS turned on and even when he did not have his VoIP ATA connected the router was still throttling his high speed data path.

For a full report check tom’s blog out at http://blog.tmcnet.com/blog/tom-keating/voip/covad-t1-speed-and-latency-test-plus-overview.asp

Interested in leasing a T1 line?  Read all about it on our site and request a free T1 quote.

http://www.whichvoip.com/index_business.html

Andy

Although problems with VoIP and digital internet phones are not as common as the likes of Verizon would have you believe, they do exist. One such problem that I have the unfortunate pleasure of encountering is intermittent or choppy audio when using my VoIP Phone. 

If you have read any of the blogs in our Techie corner you will have a good understanding of how VoIP works. With a VoIP phone call your voice is digitally sampled and divided into small sections. Each section is sequentially sent out over the internet embedded within an IP (Internet Protocol) Packet. The choppy or intermittent voice comes from one or more of these packets being lost, not sent or delayed in the network somewhere.

So what can you do if you have this problem with your own VoIP phone?

There are a few simple things that you can do before you phone your providers customer service line.

Try and characterize the intermittent voice. Is it in one direction only? Is it when you and your family are using the internet? Typically if you are on your VoIP phone it will be the person on the other end who can’t hear you properly and you will not experience the intermittent audio. A common cause of this is a lack of bandwidth in the uplink direction. Residential broadband internet such as DSL and cable are asymmetrical in nature. Comcast may promise 6Mbits, and they do deliver on their promise, however that is in the download direction i.e. to your PC or Internet phone. In the upload direction you won’t see more than 384Kbits. Try running a speed test when you are on the phone and experiencing problems and take a note of the bandwidth you are getting in the uplink and downlink directions. Compare it to when you are not on the phone.

Another simple test you can do is a ping test. Open a DOS command prompt on your PC and type the following command “ping -n 100 yahoo.com” (don’t type the “”). Below is a screen shot of what you would expect to see.

What you are doing with this test is sending IP packets to yahoo.com. The servers that host the yahoo website are sending a response back to your PC. The 66.94.234.13 is the IP address of the server on the yahoo side. The bytes are how many bytes in the IP packet and the time is the round trip time for the message to be sent from your PC to Yahoo and then sent back again. Once the test has completed you get a summary of the results, the important one is the “lost” statistic, ideally this would be 0%. Any more than about 3% and the dropped packets will result in intermittent voice over your broadband phone. Try this test a few times and see if there is any correlation between using the phone, time of day or kids home from school and downloading music!!

It’s also important to check your setup and make sure you have followed the instructions your VoIP service provider sent with the ATA. This is particularly crucial if you have a router. Your ATA is effectively a VoIP router, it will give your voice packets priority over the data packets from your PC. This means that your voice packets will be sent out on time and will not be delayed even if you are downloading or uploading large files. The most common set up is is to have your modem, then your ATA and then your router.

Try the speed test and the ping tests with the ATA removed and again try and see any correlation.

If you see a slight drop in bandwidth or one or two more dropped packets while you are on the phone then this is actually a good sign. This means that your ATA is giving priority to your broadband phone at the expense of a few dropped packets of data. Data can be resent without a loss of content unfortunately voice can not.  

These are all fairly simple test to run and you don’t necessarily have to take any actions based on the results however it will give the customer service technician you eventually call a lot of valuable information and it will save you running these tests while he’s on the phone as he’s “scripted” to ask you. Very often the technician can change your ATA settings to use a CODEC with higher voice compression thus using less bandwidth and resulting in less lost packets. They may also get you to call your broadband provider and have them check your connection. The above tests are also valuable as your broadband provider customer representative will no doubt ask you run them.

The one thing that I find somewhat ironic in the residential VoIP market is the fact that the likes of Verizon push their POTS telophony on reliability and suggest that VoIP is not so reliable. Probably half the problems associated with VoIP are problems with a users DSL or Broadband connection and not the actual VoIP application.

There are many triple play deals out there, VoIP, Broadband and TV all from the same provider like Comcast, these are decent deals, but you could find a cheaper service if you shop around for individual services. However having one company to contact when you have problems and eliminating the excuse “call the other guy it’s his problem” could be worth the extra cost.

Calum

 http://www.whichvoip.com

Technorati Tags: Broadband Phone, VoIP, Choppy Audio, Intermittent Audio, Broadband bandwidth

Voice over Wireless LAN, or VoWLAN as it is referred to, has been getting a lot of publicity recently.

What is VoWLAN?  In simplistic terms this is VoIP but over WiFi or Wireless IP.  Many households have WiFi these days just due to the convenience of accessing the Internet, especially those with laptops.  Sit in the living room watching the basketball game on the TV and surfing the web and with no CAT5 Ethernet cable or the likes getting tangled up with your bottle of beer.

How nice would it be if you had the same flexibility with your telephone.  Yes with cordless phones we have this already….well kind of.  You still need to have the base connected to the VoIP ATA or the POTS line if you have the old fashioned form of telephony   Yes the technology up until now has definitely been of the wired variety.

VoWLAN gives you additional freedom.  Here your WiFi enabled VoIP phone will talk directly do your WiFi router wirelessly.  Many providers are now releasing such phones in the marketplace or frantically designing one to release imminently as this is likely to be a huge market over the next year.

However, there are some technical drawbacks with this technology that should be brought to your attention.  All of these can be overcome but may be of interest to you.

The fundamental problem is the way that the packets are handled.  The beauty of Internet Protocol is that the packets sent over the Internet backbone can be voice, video or data.  It does not matter, they are made up of packets of digital bits and sent across the Internet to their destination.  However, all packets are not the same.  Data packets such as web browsing tend to be large packets often over 1000 bytes at a time.  Voice packets tend to be significantly smaller, often as small as 30 bytes depending on the codec used for the voice compression.

Now with the various OSI stack layers used to send data and voice over the Internet, each layer adds overhead packets that define protocols, source and destination addressing, checksums for error correction and the likes.  When the actual voice packets are so small to begin with this ends up making the data transmission very inefficient due to the extra overhead, often as bad as 30% (e.g. 100 byte packet but only 30 bytes of actual voice information). 

Why is this such a bad thing?  Well, the WiFi Media Access layer tries to give each station on the network equal access and gives no consideration to the time each station has during its access time.  Therefore, because the voice packets are so small compared to the data packets a lot of time is spent by the VoWLAN phones backing off and waiting to transmit the next voice packet.  Now VoIP is very much reliant on low latency and low jitter.  This environment does not suit it well at all especially when many Data nodes are trying to access the WLAN at the same time.  The result, if not managed properly, is reduced system capacity and poor voice quality.

Another issue is the distance from the WiFi access point.  The further away you are the lower the signal strength and the lower the transmission rate needs to be otherwise the signal to noise ratio takes a serious hit and some of the voice packets can be lost.  The lower the transmission rate the longer it takes to send the packet and the more hold-offs required by the stations on the WLAN.  Again the voice quality can suffer and of course the system capacity.

It’s not all bad news.  The VoWLAN providers have worked hard at resolving these issues or at least limiting their effects.  The upside is the lack of wires and equipment needed in your home and the portability of your voice and data network.

I hope this has given you some insight into the world of VoWLAN. 

Interested in buying a WiFi VoIP phone, check out our hardware section for more details:

http://www.whichvoip.com/voip/voip_business_hardware.htm

Any of you using this technology at home or at work now - we would love to hear about your experiences, please add a comment below?

Andy

Technorati Tags: VoWLAN, Wireless IP, Wireless VoIP, VoIP

You may have heard the new buzz word floating around the technology websites recently - Internet Protocol TV or IPTV for short.  This really is starting to get interesting.

Today Sony announced that it would be equipping most of its new High Definition TVs with an attachable module that would allow streaming broadband High Definition TV to be displayed on their TV sets, simply by pushing a button on their remote control.

Of course you will need to do some serious upgrades to your broadband service to facilitate the extra bandwidth needed for IPTV.  AT&T’s U-verse IPTV service gives customers a 20Mb/s link to their homes.  This provides high enough bandwidth for high speed Internet, VoIP and IPTV service. 

Verizon of course have their own elaborate plans to capture this market and as a response to the 1000 customers per day they are estimated to be losing to cable customers who already have triple-play.  They are, as we speak, rolling out a $20 billion investment throughout the U.S on a state of the art fiber optic infrastructure that will support Data, Voice and IPTV. 

The competition for this triple play is fierce and will be interesting to watch over the next few years.  The other question, where does this leave the likes of Vonage?

What’s your thoughts on this, we’d love to hear from you.

Andy

Technorati Tags: IPTV, Triple Play, VoIP, Internet TV

No matter what type of T1 line you ask for its bandwidth, physical medium, etc we have disucssed before will be the same. However what type of data or how the T1 is used  maybe different.

Primary Rate Interface or PRI T1 as it is more commonly know as, is probably the most flexable T1 line available. Remember a T1 is made up of 24 individual channels called DS0’s. With a PRI type interface 23 of these channels, often referred to as B(Bearer) channels, are available for data or voice and the 24th channel, the D(Data) channel is used for signaling. Anyone familiar with ISDN will recognize the terminology.  This Data channel can dynamically set up any of the other 23 Bearer channels to carry any sort of data. So one PRI T1 can carry voice, data and then if say you have to have a video conference then 6 channels can be grouped together to carry the video and voice.

This flexibility can be a great solution for small companies that don’t want to purchase multiple T1’s for different solutions. Just fill out this simple form for a free primary rate T1 quote and see the benefits a PRI T1 can offer your business.

WhichVoIP Team

http://www.whichvoip.com

The ITU define the acceptable delay for speech in G.114. The aim in any network is to minimize this delay, however this can be problematic as the designers of networks have no idea who or where you will be calling. You maybe calling from your IP phone in New York talking to your buddy who is on his cell phone in Asia somewhere - who knows how many different networks your call is being routed over. One of the fixed delays that can be controlled is the delay through the vocoder(voice compression algorithm).  

The higher the compression rate then the larger block of data the compression algorithm needs. If we look at one of the popular algorithms used, LD-CELP, Standard ITU-T G.728(Code Excited Linear Prediction), G728, this algorithm compresses a 64Kb per second sample down to a rate of 16Kb per second. Lets assume it requires 160 bytes of voice samples before the algorithm starts to compress the data. 160 voice samples at 125us sample rate equates to 20ms to just store enough samples for the vocoder to start. It typically takes 5ms to process those 160 bytes and produce the compressed data to be sent. The 160 bytes have now been compressed to just 40 bytes. By the time we add the overhead of framing information, Forward Error Correction(FEC) and the overhead of 24 bytes for the IP Packet Header then we have to send on average about 80 bytes every 20ms. That’s the same as 4000 bytes a second or 32Kbits  per second.

If we increased the time between packets from 20ms to 40ms then the bandwidth required would drop to about 24Kbits per second. (120 bytes per 40ms). However the delay or latency has increased by 20ms.

For most residential users whether 24Kbits or 32Kbits are used is not a big deal, it won’t impact them surfing the web or downloading videos, songs etc. However this 25% reduction in bandwidth effectively means that businesses could gain 25% of bandwidth translating into more simultaneous phone calls per T1 and hence less T1’s required - this could be a substantial saving.

Various new voice compression algorithms are being developed. One of the most common that’s now being used compresses voice to 8Kbits per second. Giving even more calls per T1 installed.

To see the benefits your business could benefit from switching to T1’s or VoIP visit here and fill out the simple form for a free no obligation quote.

Whichvoip Team

http://whichvoip.com

Is there a difference between a data transfer and a VoIP call? There isn’t a significant difference, the same IP packets are used, the same medium is used but there is one important difference - how big can a packet be?

With a data transfer you don’t care about latency i.e. how long it takes for the packet to be generated, then transmitted and then deciphered and re-assembled at the other end. Whether an email takes 1 second to be sent or if it takes 1 minute do we really care? With voice this latency is a big problem, this was one of the main problems VoIP faced. However over the last 5 or 10 years this problem has all but been solved. The advancements in silicon technology by shrinking the geometry of transistors enables more of them to integrated on a piece of silicon. This then enables DSP’s, Codecs, network processors(all key elements of the internet) to run faster, have more functions and have more on silicon memory and hence do more in a shorter period of time. Without these powerful key elements we would not be able to meet the strict latency requirements for a conversation or have the bandwidth we enjoy with the likes of comcast and verizon.

With a traditional phone call, 8 bits are generated every frame, they are sent every frame, decoded at the other end every frame and tarnslated back to analog every frame, it is a very linear process in that each function has the same job to do in the same amount of time.

Now looking back at the IP packet in the previous blog there is at least 24 bytes in a header before you even get to the data. So to send 8 bits every 125us(8KHz), would require at least 25 bytes - this is more DS0’s than a T1 has and is a complete waste of bandwidth. There are two main techniques used to reduce the bandwidth required, however they negatively impact the latency. As in all engineering solutions there is a trade off from what one desires with the effects of getting what one wants.

1. To reduce the bandwidth of the actual 8 bit voice sample a technique called voice compression is used. There are many compression algorithms available, these typically run on DSP’s or ASICS inside your ATA or IP phones. These algorithms take in numerous samples and through a complex mathematical process can compress these 64kKbits per second samples down to an 8kKbits per second sample or even lower. The advancements in silicon technology allow these powerful algorithms to be run ever faster and allow even more complex algorithms to be developed that reduce the final data rate even more.
2. Now for the obvious one - Send as much data in one packet as possible. This reduces the amount of overhead as the ratio of data to header information increases.

The trick now is to pack as many of these compressed samples into an IP packet but yet keep the latency low enough so as to not make a conversation impossible. Next time we’ll look more closely at the tolerable delay and IP packet frequency. 

WhichVoIP team

http://www.whichvoip.com

An IP packet is simply a the data you want transferred encapsulated in a frame or packet. The packet is typically broken up into smaller chunks to be transmitted over a T1 line. It’s kind of like writing a letter, you could take one line at a time and mail it in separate envelopes, if you lose one envelope then no big deal you could probably understand the message or at least decipher it after you received the rest.

The IP packet can be a rather complex entity with header information such as IP address sent from, destination IP address, header checksums, Identification field, protocol used and many others. Below is pictorial view of a packet. The data itself may also have sending and destination addresses, checksums, flags, etc so deciphering a packet can be like peeling an onion, revealing layer after layer.

Version - static 4 bit value indicating which version of IP is being used.

IHL IP Header Length - 4 bit value indicating the number of 32-bit words that make up the header. 

Type of Service - 8 bit value that is used tell the network how to treat the IP packet. These bits are generally used to indicate the Quality of Service (QoS) for the IP Packet.

Packet Length - 16 bit value indicating the size of the IP Packet in terms of bytes. This gives a maximum packet size of 65536 bytes.

Identification - 16 bit field used for reassembling the packet at the destination.

Flags - 3 bits indicating to network equipment if the IP packet can be further fragmented or not and if the packet is the last fragment or not of a larger transfer.

Fragment offset - 13 bit value used in the reassembly process at the destination.

Time to Live - 8 bit value telling the network how long an IP packet can exist in a network before it is destroyed.

Protocol - 8 bit value used to indicate the type of protocol being used (TCP, UDP, etc).

Header checksum - 16 bit 1’s compliment value designed to indicate errors in the header only. Every node in the network has to check and re-insert a new checksum as the header changes at every node (TTL value is decremented)

Source address - 32 bit value representing the IP address of the sender of the IP packet.

Destination address - 32 bit value representing the IP address of the packets final destination.

Options - variable bit field representing options The actual options requested are not optional in that there are well defined options that can be requested, what is optional is whether to request theses options or not.

Padding -  Variable size bit field. These bits are used to ensure a 32 bit boundary for the header is achieved.

Next time we’ll highlight the main difference between a data transfer and a VoIP call over a network.

WhichVoIP Team   

http://www.whichvoip.com

First let me state there are no physical differences between a T1 for Voice and a T1 for data. They use the same physical medium, they have the same bandwidth, the same signal coding and the same frame format. The only difference is that it is now data instead of voice.

We’ll confine our discussion to data sent via the internet.

First lets recap on the T1 frame. One frame every 8KHz, 24 DSO’s in a frame and 8 bits in a DS0. This can be seen in the diagram below. If we are talking about a T1 to a PBX then each DS0 carries one voice call, with that call occupying the same DS0 every frame until the call is terminated. 

With data transfers the DS0’s in a frame can be grouped together. If for example we grouped 6 DS0’s together then we would have a bandwidth of 384Kbits per second. This kind of generates it’s own frame inside the T1 frame 

Now we have created a second sub frame how do we know when one of these sub frames starts or stops? This is where we need to discuss IP Packets.

Next time round we’ll look at these IP packets and how they are formatted.

WhichVoIP Team 

http://www.whichvoip.com

The previous blog explained why we have 8 bits in a DS0, here we’ll explain how we get these 8 bits.

When  your voice is digitally sampled the amplitude or how loud you are speaking is captured, this happens 8000 times a second. However this digital sample is not what is sent. For a T1 line an algorithm called u-law is used to compand the data. This is a logarithmic type of algorithm that has the effect of giving greater resolution to lower value samples i.e. when you are talking quietly, and less resolution to higher amplitude samples. This intuitively makes sense, if someone is shouting then you can afford to loose a little differentiation between sounds but when someone is whispering you want to hear or accentuate the different levels so you can make out what they say. Typically when your voice is sampled, an Analog to Digital converter that has a 13 bit output is used, the u-law algorithm as well as performing this logarithmic compression has the effect of reducing the voice sample to the 8 bits that is sent.

Up until now we have confined our discussions to voice T1’s. Next time we’ll take a look at how data is transmitted via a T1 line.

WhichVoIP team

http://www.whichvoip.com

How did the 8KHz frame rate and the 8 bits per DS0 come about?

I’ll take the easy one first - Why 8 bits per DS0? For digital or telecom engineers 8 or any multiple of 2 is a perfect number. It’s very easy in digital logic to count in multiples of two thus it is very easy to extract these DS0’s, generate the frequencies and clocks required to receive and transmit at these rates and multiplex these T1 streams into T3 streams or higher rates. The trickier question is how do we get our voice to be transferred from one point to another in only 8 bits every 125us(this is the equivalent time of an 8KHz frame rate).

There is a theorem called Shannons theorem or sometimes referred to as Nyquist-Shannons theorem that drives the fundametal basics of all digital communications.  First remember that T1’s were invented long before the advent of the internet or VoIP and they were designed with the specific goal of transferring voice in a digital format. The result of Shannons theorem is that to sample an analog signal(your voice in this case), convert it to a digital format and then reconstitute it as an analog signal(your voice at  the other end) that has lost no or little information(the person you are talking to can understand you) the original analog signal must be sampled at a rate of at least twice the highest frequency you want to transmit.  The typical human voice has a frequency range of about 50Hz to 3500Hz. So if you want to sample that voice and digitize it you have to sample it at a rate of at least 7000Hz. As in all good designs there has to be some design margin, often referred to as a guard band in communication theory, a rate of 8000Hz was chosen. The frame rate of 8KHz became the standard.

Now we know why 8 bits is used in a DS0 and the rate it must be sent at. In the next blog we’ll go into a little more detail on how the 8 bits are generated and what they represent.

WhichVoIP Team

http://www.whichvoip.com

There has been a lot of publicity recently on Asterisk for your business telephony needs. 

What is Asterisk?

Asterisk is known as the open source IP PBX.  Basically, rather than using traditional PBX equipment or Cisco VoIP platform equipment, such as their Call Manager IP PBX, some businesses and universities are utilizing Linux based servers to handle their telephone calls using VoIP.  No expensive PBX equipment needed and no Cisco license fees as part of their call manager network.

The linux servers are normally found in such environments anyway in order to handle the data network for a facility so adding voice is a case of installing asterisk and connecting your phones to the server.  Analog phones will need some form of conversion to digital (e.g. using Cisco VGC gateway devices), but if you have IP phones, simple, just connect the Ethernet cables from the phone to the server.  The voice traffic will simply follow the same path as the data does now over the T1 interfaces. 

There is some downside of course.  The technical support has gone, although so has the expense   So long as you are comfortable running your voice and data network with Linux servers and have some expertise in this area so you can handle the administration, the cost savings are substantial.  Recent University studies suggest that moving from a Cisco call manager license based system to Asterisk can reduce your telephone costs by over 65%.

Asterisk is definitely one for us all to watch.
http://www.asterisk.org/

Andy

If you have ever searched for Business VoIP or have been serious about using any business telecom service then you will have come across the term T1. Over the course of the next few weeks we’ll endeavour to explain what a T1 is and delve down into the details of it for you techies out there.

So what is a T1? Well a T1 really defines 4 things.

1. The physical medium used.
2. The bandwidth or clock rate.
3. The type of signaling and encoding used.
4. The format or frame the data is transferred in. 

With a T1 there are two physical mediums. One is 100 Ohm shielded twisted pair and the other is 75 Ohm Coaxial cable. Typically inside patch panels the 100 Ohm twisted pair is used as it is cheap and flexible but for transmission to the outside world a balun is used to match to 75 Ohm Coaxial cable which is typically used to connect to the ISP or PBX.

The bandwidth of a T1 is 1.544Mbits/sec that’s for one direction, one of the advantages of a T1 is that you get this dedicated bandwidth in both the uplink and downlink directions.

T1 uses a signal encoding called B8ZS which stands for Binary 8-zero Substitution - quite a mouthful. Without going into too much detail just now, this type of encoding allows data and a clock to be sent across the same cable - vital for synchronous data transfer.

Finally the format - Each T1 is split into 24 channels referred to DS0’s. Each channel is 8 bits wide and the frame is sent on a 8KHz boundary. The data capacity is therefore 24×8x8000 = 1.536Mbits/sec. Didn’t I say earlier that a T1 was 1.544Mbits/sec? Well there is something called a framing bit, this is 1 bit every frame that a receiver on the other end can lock onto to ensure that whatever equipment is there can extract the correct DS0’s in the correct order. This framing bit makes for an extra 8000 bits/sec giving a grand total of 1.544Mbits/sec.

T1’s are essential a for any business that employs more than about ten people at the one site. They are very reliable and give you a guaranteed bandwidth.

If you are starting a business or you already own a  business and are planning on expanding, fill out this form for a quote on how much a T1 is - you maybe pleasantly surprised.

Next week or when we have time we’ll explain how the 8KHz frame rate came about and how we get 8 bits in a DS0.

WhichVoIP team

http://www.whichvoip.com