WhichVoIP Blog

Today in New York Microsoft’s Steve Ballmer and Nortel’s Mike Zafirovski provided an update of the of their partnership, originally announced in July.

The full Webcast and transcript is available at http://www.microsoft.com/Presspass/exec/steve/2007/01-17Nortel.mspx, but the key announcements are as follows:

1. Integration between Microsoft’s recently released Exchange Unified Messaging and Nortel’s IP-PBXs
2. Extending the combination of Nortel’s Converged Office, which integrates with Microsoft Live Communications Server and Nortel CS1000 IP-PBX, to now be available upon CS2000. This takes the solution from the small and medium enterprise, to the larger enterprise
3. Integration of Nortel’s Multimedia Conferencing and Microsoft Office Communication Server
4. Creation of a solution targeted for deployment within a branch office.

This announcement has been highly anticipated, as the original announcement in July was very weak on actual details. With this update, Microsoft and Nortel have shown that they have made progress over the last months that provide real solutions and value to the end user.

Though there is some time before these solutions become available, the discussion clearly shows the benefits of migrating from “old telecom” to “new IP technologies”. In this case, the ability to have a single unified client (Microsoft Office Communicator) shows real value.

Technorati Tags: Nortel, Microsoft, Unified Messaging, Unified Communications, VoIP

A lot of the negative reviews we get all have one thing in common, actually it’s 2; bad customer service seems to be a common complaint but another is choppy or intermittent audio during a phone call. Over the past week or so I have had the same problem with my VoIP service. I’m still working on it!!

So what causes it? A lack of bandwidth is one cause. Run this speed test and make sure you have at least a 200Kbs in the uplink and downlink direction. Another cause which can be related to lack of bandwidth is packet loss, every time a packet is lost in the network you are loosing some audio. How can you tell what your packet loss is? - well type the following command in a DOS or command window on your PC “ping -n 100 yahoo.com” don’t type the quotes. This command sends a small packet of information to yahoo.com servers and they send a reply saying they got it. The -n 100 just runs the same test 100 times. Try it a few times and see the results. If you are not using your VoIP phone you shouldn’t see any more than about a 1% packet loss. More than about 3% to 5% and the audio drop out becomes very noticeable.

A lot of problems users of VoIP have is that they don’t follow the setup instructions they get - I know because I have helped so many friends read their instructions . It is important to put your ATA directly after your modem and then connect you PC or router to the ATA. Your ATA probably has Quality of Service (QoS) built into it, in that it gives priority to your voice  - which is what you want.

If you type the ping command at your command terminal while you are using your VoIP phone you’ll probably see an increase in packet loss - this is actually a good thing as you ATA is giving priority to the voice packets.

Not all VoIP problems are related to your internet however a good portion of them are, unfortunately you suffer and the VoIP company gets the blame. So, before cancelling your VoIP service test yout internet connection. If it is bad then you’ll just have the same problems with your next VoIP provider.

I’ll let you know the cause of mine when I Know.

 Calum

http://www.whichvoip.com

Following on from the recent Microsoft announcement of the Office Communications Server public Beta, is a Microsoft Press Pass Q&A article with Anoop Gupta - Vice President for Microsoft’s Unified Communications Group. In this article, more information is available on the vision and strategy within Microsoft that makes some interesting reading and explains how Office Communications Server will be the key infrastructure piece within Microsoft’s push into Enterprise VoIP.

The simple answer is - if you are approaching 8 T1 lines then you should look into a DS3 connection.

8 T1’s give you 12.352Mbits of bandwidth a DS3 gives you 44.736Mbits of bandwidth(equivalent of 28 T1 lines), the cost however of 8 separate T1’s fast approaches the cost of 1 DS3.  DS3 connections like T1 connections are exceptionally reliable, however if you move from multiple T1 lines to a DS3 it is worth having a PRI T1 as backup incase something happens to your DS3.

What can you expect to pay for a DS3?  Just like a T1 the cost varies mainly based on distance from the Central Office, so anywhere between $3000-$5000 a month.

If 44Mbits is a too much bandwidth - if there is such a thing  there are fractional DS3 connections just like fractional T1. Fractional DS3 lines are probably more common than full DS3 connections. Not only is the cost a factor when migrating form multiple T1 lines to fractional DS3 but fractional DS3 also has an advantage in that it lends itself nicely to upgrading your bandwidth as required, since no extra cables or installation has to be done. The cost of a fractional DS3 typically starts at $2000 probably a lot less than the cost of 8 T1’s and probably approaching the cost of 4 T1’s.

Get a free DS3 or fractional DS3 quote and see how you can reduce the cost of your communication needs.

Whichvoip Team

http://www.whichvoip.com/

As we’ve stated many times before a T1 line physically looks the same in terms of bandwidth, frame rate, clock rate, etc, regardless of what type of information it ultimately transports. So this being the case can’t you just get a T1 and use it for what ever you want? Unfortunately not. We have talked about PRI T1 previously and if you can get this for a small premium over a voice T1 or a Data T1 then that is by far the most flexible solution.

So what is the difference between a Voice T1 and a Data T1?  The only major difference is what the T1 is terminated by at the Central Office or switch. If you want a voice T1 then the provider will physically connect your T1 line to a voice only Line card in a switch such as a 5ESS that Lucent Technologies sell in the US. If you need only data then you maybe connected to a physically different piece of equipment such as a Cisco 7500. These cards and equipment are simpler and have less software than a PRI T1 termination line card, hence Voice or Data T1 lines tend to be cheaper but of course you loose your flexibility.

Obviously as user you don’t really care what is at the other end of the T1 line but it is important for you know what sort of T1 you need and want. Let the WhichVoIP team help you get the solution that’s best for your business - Free T1 quote.

WhichVoIP Team

http://www.whichvoip.com/

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

Microsoft recently announced the public beta of Office Communications Server 2007. This server software is Microsoft’s first entry into the Enterprise VoIP market space, and is a reasonable initial version.

Adding voice and conferencing to the successful Instant Messaging and Presence solution, with the previously named Live Communications Server 2005, is a move to enable Microsoft to deliver it’s Unified Communications vision. This vision does not only include voice, but conferencing, presence, data and text communications, within one integrated client (Office Communicator). With such a client ,the user is able to not only replace their old TDM phone (as with any softphone) but be able to deliver additional benefits via other forms of communication, and the improvements provided by use of presence.

The server solution, is a scalable solution for both small and large enterprises, and completements a PBX solution within a branch or headquarters environment.

Watch this space for additional information, and a review of the overall solution

Over the past few weeks we have gone into some depth as to what T1 lines are in terms of voice T1, sampling rates, frame width, data T1 lines, IP packets, voice compression, packet delay, etc. But what do you need for your business?

Over the coming weeks I’ll outline a few different T1 Lines from fractional T1 lines to full rate T1 lines to primary rate T1 lines to data T1 lines and why they are important for you to know about.

In an earlier blog Tony talked about the different costs for T1 lines. The important point of that blog was to ensure that you get more than one quote. The cost of T1 lines are completely dependant on the distance to the central office. A T1 cost in say Seattle can vary from $350-$600 depending on whether you contact Speakeasy or say Bandwidth.com. Don’t do this work yourself let the experts help.

Fractional T1 lines 

Fractional T1 lines are as their name implies portions of a T1 line. You only get a fraction of the possible 1.544Mbits per second bandwidth. Common fractions are 765KBits per second, 384 Kbits per second and 128Kbits per second. Why would you get a fractional T1? Well in the past, before the year 2000, it would have been cost - Not much of a problem though in the dot com boom years The less bandwidth you want the less you pay. However since the boom years of the late 1990’s there is a lot of infrastructure in place that is not being used and hence not making companies any money. A glut of bandwidth is good for you. The differential in cost from a fractional T1 to a full T1 is probably no more than about 15%. So for 15% more money you get twice the bandwidth. For this reason Fractional T1’s are not that common these days.

However if you have lots of small offices dotted around the country then this 15% saving could add up. But again don’t do that work yourself let the experts find you the best price.  

Whichvoip Team 

http://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

We have talked a lot about what a T1 is and that the benefits are the increased available bandwidth that it gives your business over say, a buiness class DSL service. But where does it fit in on the bandwidth scale for different bandwidth offerings and how much does a T1 circuit typically cost. Well first of all, here is a brief summary of various bandwidth options available:

33.6 K (Modem)          33,600 bps
56 K (Modem)             56,000 bps
64 K (DS-0)                64,000 bps
128 K (ISDN)             128,000 bps
256 K (DSL)               256,000 bps
640 K (DSL/Cable)     640,000 bps
768 K (DSL/Cable)     768,000 bps
T1, DS-1                     1.544 Mbps
T3, DS-3                     44.736 Mbps
OC-1                           51.840 Mbps
OC-3                           155.520 Mbps
OC-12                         622.080 Mbps
OC-48                         2.488 Gbps
OC-192                       10 Gbps

As you can see, the T1 is the next step up from DSL and they are available as both full and franctional services.

So how do you get a T1? Well unfortunately the process is not quite as simple as getting your DSL going. The first thing you should do is get one or two quotes from service providers. To help you with this, WhichVoIP has partnered with some of the best providers in the business to provide you with competitive quotes. We have them squaring of against each other for your business Just go to our Business Solutions section and fill out a short request. Simple eh!

But then how long does it take? Well after you are happy with the deal you are getting it will take around 25 to 50 days to actually get your install completed. So plan ahead. This is mainly gated by the local exchange carrier who needs to coordinate the installation.

How much should I pay? Typical costs for a T1 circuit range from around $350 to as high as $650 per month. The reason for the discrepency is that the last mile from the Central Office to your premise has a high weighting in this calculation. So if you are close to the Exchanges Central Office, you could be in luck. However, the best way to make sure you are getting the best deal, is the good old method of getting more than one quote. This is where we hope we can help.

What if I still have questions? Then send us an email at customerservice@whichvoip.com and we will try to assist you.

In summary, small to medium sized businesses should never need anything more than a T1 or multiple T1’s. Some businesses prefer to get 2 T1’s for load-balancing and redundancy, especially if it is running VoIP on its network.

Cheers

Tony

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