PBX 2 SIP

It sure does, and Jitter can be a very nasty thing in VOIP.  But don’t forget about dropped packets and their negative affect on the network.

So, you understand these problems, but how do you find them, how do you measure them, and how do you monitor them?  There are a ton of products on the market, and most are pretty pricey.  We’ve talked about the benefits of Wire Shark, which are hard to beat.  However, there is another one to look at as well.  This one will take a small investment, but it is well worth the cost.

The product is Trace Buster.   Okay, not my pick for a product name, but I do like the product, and it has helped me bust some traces that had me scratching my head.  For $99, trace buster gives you an application that grabs your packets using Wire Shark, and then displays it in very easy to read graphs that show you what level of jitter you have on each call, how many dropped packets on a per call basis, and it will graph out all of your traffic by protocol, providing detailed information on audio and video calls.

It is the bomb!  Yep, pick up the phone, call your carrier and tell them that you have average jitter of xx, and a maximum of xx.  Let them know you are seeing x% of dropped packets coming from their network.  Let them scramble to figure that one out for themselves – and you will have a happy customer.

So, you don’t have the trusty old butt set to do your troubleshooting for you with VOIP, so what’s a phone guy/gal to

 do?  First thing is some planning.  When you put in a VOIP systems, it’s important that you build yourself some test points in the network, so you can insert, or extract data to do troubleshooting.  Remember, its not as simple as pressing the bed of nails on the wire and listening anymore.

But, there are some other need tricks that you can do.  For example.  Ethernet switchs have a feature called Port Mirroring.  Make sure the switches you install have Port Mirroring.  Here is how it works.  Lets say a phone you are having problems with sits on port 20 of your ethernet switch.  And, lets say that you have your laptop on port 24.  In the switch, you port mirror port 20 to port 24.  Now, every packet to and from port 20 will also be sent to port 24, where you can see it on your laptop.

This is important, because on your laptop, you are going to install your newest best friend in networking – Wireshark.  Wireshark is a packet capture program that grabs every packet on the line, and classifies them for you by protocol, IP address, port, you name it.

So, getting back to that telephone with issues.  What you want to do it start wireshark and start capturing packets on the phone with issues.  When wireshark opens, go to the Telephony tap on the menu, and scroll down to VOIP and click on that option.  IT will open a new box that will allow you to follow the VOIP calls on the system.  Now place a call to reproduce the issue you are trying to resolve.

Your call will appear in the Telephony/VOIP box, and you can select it and graph it, which shows the session and each step of the call setup and teardown, and you can also listen to the audio portion of the call as well, either incoming, outgoing, or both.

Clicking on a record in the graphic mode will open the actual packet from the trace to let you drill down further for details.  This is a great opportunity to see exactly what your call is doing or not doing.  Don’t forget, you can trace another port that doesn’t have the issue to see what a good call looks like as well, so you can do a stare and compare of the good call versus the bad call.

This is a great tool, that can save you a ton of lost hours troubleshooting on VOIP systems.

Althought even that is probably not accurate, since the data is from January, 2009.  A full year later, I suspect we are looking at even greater numbers.  But even still, 1 out of every 5 PBX’s on the market today is Open Source.  That is incredible.  Read the article from No Jitter

The data also points out that its not the large integration companies that are providing the open source products, but the smaller companies.  Go figure, businesses are finding the lower cost of equipment, the lower cost of labor, and the highly responsive and reliable small business telephone companies are delivering the goods.

“For Open Source to be the success it is, might suggest the largest VARs were early to seize a nascent business opportunity, and could justifiably take credit. That’s not the case. Working with a list of the largest 500 VARs, we began one series of interviews working our way down the list from the top, only to conclude that smaller VARs deserve more credit for the growth of Open Source. So far, large VARs have been largely under-accounted-for in the Open Source PBX market.”

This is great news for the small mom and pop shops out there that service small and medium business’s from their home.  Your business is intact, as long as you grow and keep pace with it.  This web site is here to help you stay up with the Open Source Market.  Send us an email and see how we can help you transition your current legacy PBX customer to a new Open Source system that is cost effective, reliable, and robust.  It will give your carreer a new lease on life.

You can’t argue with that logic, you simply plug the butt set in at different places in your telephone network to check for service.  BTW, a butt set is a test phone that was normally carried around by a telephone technician for testing.  Clipped to a tool belt, it usually was swinging on the backside of the belt, near the……well you get it.

Today, a technician needs the ability to do testing like that to see where issues are in the network, and you can do this with the right equipment.  Let me give you an example. You have a network with a Router, a Firewall, a SIP server, and computers and phones.  We’ll use a small network as our sample, but it will scale to larger networks as well.

For this solution, we are going to use a 48port POE Switch.  In my example, I used an HP 2610, but you can use your preferred switch.  This particular switch supports vlan tagging, layer 3 switching, and most importantly, port mirroring.  Port mirroring allows you to direct a copy (mirror image) of all packets on a port to another port so you can take the mirrored traffic to view in real time.

What we will do is take ports 1 and 2 of the switch and put them into their own untagged vlan, we will call it vlan 31.  Our Internet connection will plug into port 1.  Port 2 will be connected to the WAN side of our router.

Next we will take ports 3 and 4 and put them into their own vlan as well.  We will cal this vlan 37.  The Lan side of our Router will connect to port 3.  Port 4 will connect to the wan side of our Firewall.

We have just created a chain through the ethernet switch that provides a test point in front of the router, behind the router, in front of the firewall, and behind the firewall.  By using port mirroring on the switch, we can direct a copy of all traffic to any of those ports to a port that our wireshard packet capture machine is on and get great troubleshooting capability at any of those ports.

By having our phones, servers, etc. on their own ports, we can drill into any of them as well to look at the specific traffic on each port. 

 It’s not a butt set, but you don’t have to carry it around either.  Yet, you can follow traffic around on your network, point by point by point.  You can simply redirect your port mirror to the port your packet capture device is on, and do custom captures as your troubleshooting of your network.

Welcome to the network Phone Guy!

It’s been the thorn in the side of deployment of VOIP to the desktop, the fact that all the cable plant needs to be upgraded to Cat 5/6 to support ethernet.  In many cases its okay, you put a two port phone there and bridge the computer off the back.  This does add some cost to the phones.  But in many cases, companies are hiring firms to update their cable plant to add more cat 5/ cat 6 cable to support their VOIP deployment.

What if there was a way to keep that old phone cable to support VOIP?  After all, the VOIP call will only require at most about 80 k of bandwidth!  There is a new product on the market that will allow you to deploy your VOIP phones over the existing cable plant.  This is a great trick that someone should have thought up years ago.

I’m sure you have had DSL at your home at one time or another.  It was your fast connection to the internet, versus your old dialup account.  And, DLS has gotten fater now, up to 25mb with standards, and they can deliver serve up to 1,800 feet away.  What if you could put that into a small appliance in your office to upgrade your existing voice circuits from analog to 25mb DSL to the desktop?  Thy is exactly what Phybridge has done!  It is so simple, you simply place their Unyphyer appliance in the rack, run a voice grade amphenol cable out of it to a punch down block, and connect it out to where your phone is today.  You plug a small adapter (dsl modem) into the jack, and connect up to 5 ethernet devices off of the end of it.  VOIP phones, computers, etc.

And, talk about concerns for QOS on the network?  The Uniphyer has a single gigabit port off of it that plugs into your network, passing on QOS and VLAN tagging to your data network.  One interfact to your data network for all of your voice.  Now you can run these two separate networks, knowing you have true QOS on the voice side.  Best of all, it installs just like a legacy voice system – easy for a legacy VOICE guy to do.

Savings from not having to rewire your entire warehouse, school, commercial business more than saves on the cost of this solution.  And the rewards are in your continued high qualify voice over IP installation.

I remember when I was young, seeing the AT&T presentation at Disneyland, telling about the video phones of the future.  That was the 60’s, and we still don’t see them in use today.  But, predictions are that 2010 will be the year that Media phones are accepted in the marketplace.  Oddly enough, it will be the consumer market that adopts it first, with non-computer users getting them for video, which fills the vast majority of the internet today.  They phones will allow them to check email, youtube, news, etc. from their Media Phones.

Of course, they will want to call their lawyer, their banker and their accountant from these phones as well.  Is your business ready for these types of calls?  Will you have to run to a spare conference room to take them?  And, is your phone system prepared to handle it today?

Chances are that if you are not converted to IP, you won’t be able to take the video call, but will speak to this customer in a traditional voice call.

Legacy phone systems just aren’t prepared for this, and can’t support the User Agents required to support Video.  But, there are solutions for this, and they are avaiable now.  Take Polycom, the number one independent manufacturer of telephones, they have their new VVX 1500 business media phones.  Its a great looking phone.  And, as a sip phone, it supports multi-media, including video.

Snom has recently released its Snom 820, their multi-media User Agent.  And Grandstream has one as well.  The beautiful thing about SIP, and multi-media, is they are designed for each other.  SIP PBX’s will in most cases carry video with no issues.  It’s just a matter of turning on the video with a push of a button and the video conference is active.

Legacy PBX technicians need to learn SIP, and they need to understand IP to get there.  Augmenting the legacy equipment with SIP is a good first start.  Replacing it with SIP standards based IP systems is the long term answer.

Okay, your a telephone technician, you don’t know networking as well as you would like to, and you have a customer that is asking about a router, a VPN, a mail server, etc.   What do you do, walk away, tell them to call someone else, or struggle learning a new product to satisfy their needs?

Today I had a great discussion with one of the engineers at Sutus.  He told me that he has seen people buy a Sutus server strictly for the VPN and email solution.  I said to myself – WHAT?  At first it seemed ridiculous.  But then, it seemed like genious.  A guy that knows telephones, but isn’t a networking guy can configure a Sutus in a flash, the user interface is so simple.  And, this simple box becomes a firewall, throw the Cisco Router out, it becomes a firewall, throw the WatchGuard out, it becomes a mail server, throw the outdated Exchange box out, it becomes a file server, throw the old clunker box you call a server out, and it is your VPN, goodbye Contivity or special router.

Whats best about this solution is its one box, easy to configure, easy for the customer to take control of, and affordable.  I’m sold on this idea, I really am.  And, for about $2,300 your ready to go with all of this in one little box.  Best part is, as a service provider you can access it remotely, configure it remotely, and even get email notification, when something is struggling, via email alerts.

I think this solution is hard to beat, and any old telephone guy can handle it.   With this product, you can pin the IT badge on your lapel.

And, when the customer is ready, you can turn up that IP telephone system in the Sutus Server, and turn off the old legacy telephone switch that the customer is trying to hold onto.  Heck, don’t even inflame them by telling them its in there, just wait till they are ready to pitch the old system, and educate them on how ready they are to turn up the new one………………..

It wasn’t long ago that Lucent rolled it’s struggling enterprise division off to create Avaya.  I remember wondering how they would survive being a single dimension company like that, high cost of manufacturing, with it’s low margins, and competitors like Cisco and Nortel knocking on every door that they knocked on.

Enter 2009, and Nortel Enterprise Division is now being swallowed up by Avaya, lock – stock – and barrel.  Who would have thought that these two telecom giants would one day be housed under the same roof.

So, what does this mean for Nortel – will they be canabalized by their old arch enemy, taking the spoils, stripping out the best of their technology, calling on their customers as they swallow up the old legacy Nortel switches to be replaced with Avaya’s antiquated technology.  Or, will the FUD masters at Cisco skim the cream of the crop off of the Nortel Customers?

There are millions of companies out there with older Nortel small business systems, ripe for being replaced with newer VOIP systems.  Maintenance creep has caught on, people are paying a premium to support these old legacy systems, and don’t even realize it since it has slowly creeped up – 3-5% a year since it was purchased.  The cost of old Legacy trunks is a drain on the Operational Expenses, and SIP trunks are waiting to help reduce expenses.  Yet, support is limited in these old legacy devices. 

Old time telephone technicians need to start thinking about newer VOIP systems, replacing the old legacy boxes with a new shiny server based system, with all the latest bells and whistles like find me, follow me, Unified messaging, Conference bridges, integrated telephone browsers, click to call, and wireless calling to their mobile devices.  It’s a bright new world of telecommunications, will you keep up with it?

We think so, drop us a line, tell us what you are wanting to learn to keep your skills current – we will include new content to help the transition from the Nortel or Avaya to new standards based products for your existing legacy customers.

SipFoundry PBX installaton services

In a great announcement to support the notion that PBX’s are not going anywhere anytime soon, UK based OnRelay has announced the General Availability of its Mobile PBX solution, Mobile MBX.  Mobile MBX is based on SipFoundry’s Sipxecs Open Source product.  This is great news for PBX technicians, and companies everywhere as it demonstrates the need for PBX, and even shows how the features offered by them are making their way into the cellular network as OnRelay provides their cellular users the same features enjoyed in most office based PBX systems.

Of course this is just the beginning of something great.  Legacy PBX’s will need to provide the same features and functionalities of the mobile phone or phase retirement.  Current PBX technicians not familiar with SIP based systems need to learn them now.  Sipfoundry is a great product to cut your teeth on, because it is feature rich, and has a great many followers.

Technicians can learn the features of SipFoundry, which runs on several flavors of Linux rather quickly, with support from local telecommunications companies that specialize in the deployment of Open Source products.  One such company is Misiu Systems, that offers installation, configuration and deployment of SipFoundry SipXecs to small and medium businesses throughout the United States.   Their unique proposal is the delivery of a fully configured PBX in a server, ready to be plugged in and cut over by a local telephone technician that doesn’t need to be an expert on the system. 

Remote configuration, support and monitoring can be performed by contract staff, while the local hands on work is completed by the same technician that has performed duties on teh corporate communications systems for years.  This model provides a great level of support, while limiting the licensing fees of many of the products on the market today.  Over time, the local technician learns the system and takes on more responsibility of the system, providing a smooth transition to the most cost effective solutions available.

This model seems ideal for todays business that is concerned about costs.  Added savings from the use of SIP trunks that terminate directly on SipFoundry rack up more savings to lower the total cost of ownership in todays VOIP PBX crazy world. 

Legacy techs – read – this is a good day to be in PBX services.  Find a partner, and move forward with your customers.

Check out this video on SUTUS, a Canadian manufacturer of an All-In-One PBX that includes VPN, ethernet POE switch, firewall, PBX, email, and file server.  Pretty incredible.  The best part of this system that is getting rave reviews is its simplicity in setting up.  Old Geezer PBX techs or Key System techs can install this easily, without training, and can create customers for life.

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The system uses Polycom phones, comes with a three port FXO media gateway to support local trunks, as well as one fax port.  It supports VOIP trunks from about 6-8 different providers, and support for the SIP trunks is included.
Autoattendant, Voicemail, Web, Ethernet Switch, POE ports, traffic shaping, VPN!  What more could a small business need to get off the ground!  Incredible.
Misiu Systems sells this product to customers and ships it configured, ready for installation and tweaking by PBX and Key System geezers.  Contact Misiu Systems at info@Misiusystems.com for a quote on this great system.  Product Info at Misiu Systems.

Thats right, for a mere $6.95 you can download a SIP softclient for your cell phone.  When you walk around the office, you can get calls on your Iphone over the wireless sip client.  Out of the office, simply connect to the office with a PPTP connection and you are connected to the office phone system.  It’s all pretty simple, and magical.

Check out how this is done with the SUTUS IP PBX, it doesn’t get much simpler than this.  http://blog.sutus.com

Know of another slick SIP application?  Tell us about it

Here is the beauty of SIP trunks.  They can be delivered over your existing Internet service.  Depending on the number of trunks you use at one time, you may have to bump up your internet service.  However, SIP trunks can be used with your legacy PBX, or they can be used with a newer IP-PBX.

The newer IP-PBX systems will take in SIP trunks in their native form as IP connections.  This generally requires no hardware, or possibly a Border Gateway if not supported in the PBX soft switch already.

Your legacy PBX can either be upgraded to support VOIP via SIP, generally not a cheap thing to do, or it can have a media gateway that supports converting your SIP trunks to Analog trunks or PRI or BRI.

The beauty of SIP is that it is ordered in increments of 1 line at a time.  A PRI is ordered 23 lines at a time. 

Most SIP trunks will cost you about $17-$22 a  month for unlimited usage, and have included free long distance, or discounted long distance as well around $.02 a minute.  Of course you can get them even less from some of the discounters, I have a trunk that cost $1.49 a month plus usage, or $7.50 a month for 3500 minutes. 

If you figure you are probably paying $45 a trunk today, you could be seeing a 40-50% savings on the cost of trunks to your business.  When you consider a IP-PBX running on a low cost server, versus the maintenance cost of your legacy PBX, is there a reason to pour more money into what you are doing today, when eventually you will have to go to IP Telephony services anyway?  VOIP is primetime, its time to start reaping the rewards with a good solution.  And, don’t only think the legacy systems are the right solution.

No question, traditional T-1 has served you well during your career.  24 voice channels of TDM, 23 if PRI, but they work, they are always there and ready to go.  Your PBX has ran well with them over the years.  And that might be their undoing as well.  Imagine if you had 24 cars that you didn’t drive regularly – its an expensive proposition.  TDM trunks are the same – they are there, you pay for them whether you use them or not.   In this economy where excess is being considered for removal, tranditional PBX trunks are the best example of waste.  IP Telephony has changed that for good with SIP trunks being the new standard for VOIP deployments.

Imagine if you only paid for them as you needed them?  When you have 5 calls, you have trunks.  10 calls, 10 trunks.  You can imagine the savings if the carrier is only charging you for what you use.  That is exactly what SIP brings to the table.  And, with its deployment across the internet, it’s easy to get a number in Los Angeles for your New York Office, at no additional cost.

Think about browsing the internet.  You open Internet Explorer, and you connect to a site.  The connection to that site is there the entire time you use it, and when you close the site it is gone.  Think of SIP trunks the same way.  The physical connect is there via an Internet connection, and the call can be established much like the internet explorer connection when a call needs to be placed.

When a call is placed, an Application level connection is made from your SIP PBX to either the Public Switched Telephone Network (PSTN), or via a SIP gateway to another SIP PBX.  The bandwidth not being used for calls is now being used for any other type of traffic – data or video.

The other nice thing about using IP for this traffic is that in reality a voice call that has a 56k or 64k channel reserved for it, is barely used by voice.  It’s a big fat pipe that is empty most of the time, even when used because voice just doesn’t fill all of that bandwidth.  This excess overhead in a SIP network become available for other traffic as well.  And, with SIP, we can use different CODECS to compress the calls.  Compression, although it adjusts the quality of the call down, it frees up bandwidth, which equates to cost savings.

From a carrier’s perspective, there are cost savings as well.  It’s not uncommon for a customer to see a savings of 30-40% when they transfer their service from traditional TDMA trunks to SIP.  The carrier’s save money and pass some of these savings on to their customers in order to remain competitive with the new companies that sell only IP Telephone like SIP trunks, Hosted PBX services, Voip Phones, etc.

The cost savings many times can help to justify the cost of SIP PBX equipment, media gateways, etc.

If you have further questions on this subject, please leave a message in our FORUM, we will be happy to help address them for you.

Very Basic PBX

So we know what a PBX and a Key system is made up of.  We have a box, with trunks or lines coming into it, and we  have telephones coming out of it.  Sometimes we have an external voicemail, or one that is built into it, and other peripheral devices possibly.

Just to clarify, this PBX drawing is very basic.  Many times the stations will be cabled out with amphenol type cables, etc which are not reflected in the drawing.

The design of new telephony networks relies on different equipment and technology, generally VOIP and IP Phones.  Gone today are the big central proprietary PBX’s that have been around for decades.  These boxes as you know were attached to phones, I won’t call them dumb phones, but limited intelligence phones., that generally only spoke with one legacy PBX.  They were not considered interchangable like todays IP Phones based on SIP.

Today, new telephone networks rely on telephone instruments that are smart agents on the IP network, able to keep a conversation alive without the central “nerve” of the network – the call server.  Imagine if you unplugged a legacy PBX during the middle of the day – all hell would break loose, but not in some of todays systems.  In a User Agent soft switch, a server sets up the signalling between the IP Phones, and then a one-to-one communications begins between these two smart devices that does not rely on a central call server to keep the call up and running.

This next drawing breaks down the legacy PBX into components that are used in newer IP based system.  We still have trunks coming into the network but they don’t terminate on a PBX box anylonger, they terminate on a Media Gateway, or they appear as SIP trunks in the form of an IP connection.  Media Gateways are just that – they are a gateway for connecting one form of media to another.  In this case, analog or digital telephone lines, to SIP based IP lines.

The PBX Box itself with its processor, operating system, etc. is replaced with a server or servers depending on the implementation.  The server in many cases includes all the features that a traditional PBX has, plus more, and can include an integrated voicemail, conference bridge, find-me follow-me, unified messaging, computer telephone integration (CTI) and everything else you would expect in a PBX.  And, it gets programmed much like the traditional PBX, with a small exception.  They are generally browser based and simpler to configure.  One of the characteristics of VOIP based systems is the old languages of the traditional PBX is thrown out with new easier to use programming interfaces that are generally browser based and simpler to work with.

As you can see, its not so difficult to understand how yesterday’s technology morphes into todays technology.  I like to relate where we are today as being similar to the day when mainframe computers and dumb terminals ruled the office.  Companies were tied to their single vendor of choice, and applications were limited.  Along came DOS, and then Windows, running on computers that could run their own applications.  Today, we are seeing the same revolution with telephone systems, where companies are no longer tied to a single vendor, their phones are not expensive proprietary phones, and Open Source products are becoming the norm for a PBX product.  IP Phones have features like Video Conferencing built right into them.

The world of IP Telephony is clashing with the world of Legacy PBX systems, but this transition to VOIP Business Systems and VOIP Business phones is a healthy change for an industry that is still clinging to its old way of doing business – phone business.

Please leave your comments along with suggestions for other discussions of Interest.

There are the three r’s we learned as a kid – reading, writing, and rithmetic.  And as old time telephone installers, we learned the 4 r’s – Ring Right Ridge Red.

In the old days, station cable had 4 conductors which were red, green, black and yellow.  The lines we installed from the pole to the house was a single pair, with rounded corners on one side, and a ridge on the other side.

Of course we all learned about Tip and Ring, the two conductors on a pair of telephone wires.

And we learned a nifty slogan to make sure we did our work correctly.  The 4 r’s.  Essentially, the red wire was always the ring, and the ring always had the ridge, and when terminated, the ring always went on the right side.  Ring Right Ridge Red.

Now I am showing my age.  But, those basics were essential to learning to do my job as a telephone installer.  Just like learning what SIP is to VOIP to  today’s installers.  SIP is the Session Initialization Protocol, and it is a framework that is designed for interoperability between systems.  It is the new technology that drives all new IP Phones and Call Managers.

Just like if you reversed the ring side of a telephone line you had problems, if you don’t adhere to the SIP standards you have problems.  If your ring is reversed, sometimes phones don’t work.  If your SIP is not compliant, sometimes phones don’t work.  Same principal, different time and technology.  But the situation is the same – we all have to talk the same language, and we all need to work together to create a system from end to end or it just doesn’t work.

Have you learned a new buzz word that helps you do your job when working with SIP?  Share it so others can benefit from it.  There are a lot of old Legacy PBX telephone systems out there relying on Ring Right, Ridge Red that need to be replaced with new SIP based IP Phones, Call managers, soft servers, etc. to get business to the next level.