Traditional Voice Networks
From the old days we have Private Branch Exchanges and these PBXs contained cards that would support the stations and the lines and the trunks that you would use to connect up this separate phone system solution. They also came in a couple of flavors. There was an analog version and there was digital version, and the digital phones would actually convert an analog voice to digitized stream and connected through to PBXs. Where the analog phone of course were just analog end-to-end. We also had voicemail solutions that would tie into the PBXs and trunks, tie trunks. These were dedicated circuits that would connect PBXs to each other or we could even use them for some of the music on hold solutions depending upon the vendor. And then there were CO Trunks, that would allow you to connect out to your local central office on a PBX and this again could be analog or digital. So sometimes there was conversions, sometimes there wasn’t. Traditionally though these were separate networks. Separate solutions from data network and then even from your video network, so you could have three separate networks setup to support your voice, your video and your data. We're going to see moving forward that we're going merge these technologies, but for now let’s still focus in on the fact that in a Voice over IP solution I might want to support some of these older technologies.
Let's say I still have a good old fashioned fax machine and I want to plug it in to my new Voice over IP Solution, what do I do? There's Foreign Exchange Stations. I think station - fax machine, single line phone, they're stations. FXS Ports can be plugged into your router and that’s what will support a single line device. FXO, Foreign Exchange Office, allows me to connect a single line from the Public Switch Telephone Network. You may still have dedicated phone lines that maybe that phone number has been there for years and you want to make sure that you can still keep that number and tie it into your solution, you can by plugging it into that FXO Port.
We also have E&M Ports, depending upon what you read, Ear and Mouth, Earth and Magneto, that’s what that stands for. This will allow us to now connect to external music on hold sources or create tie lines between our PBXs. Now you might be saying, "How do I determine if I need this?" Well if your PBX already had an E&M card in it, it already spent the money and you were using this to tie together your PBXs. Now you want to tie your PBX maybe into your Voice over IP Solution until you're ready to launch into a full IP Telephony Solution, that’s when you'd purchase the E&M connection for your router (Voice Interface Card - VIC), so could have that old PBX plug in to that E&M Port for connectivity up to your Voice over IP network.
We all know that phone lines are not enough to support a solution or a site. T1's - those are the big guys, the digital circuits of T1 connectivity that give us 24 ports (or E1's in Europe with their 30 channels) to allow phone calls to come into our system, so we can purchase T1 cards. For a small office we could get a BRI connection. The BRI gives you two channels for voice or data and then it gives you a D-channel and that D-Channel is data about whatever call is taking place, so if it’s a data call or a voice or a video call down one of those other channels all the information about that call setup traffic, etc goes down that D-Channel. So with this digital circuitry we have the ability to choose on a T1 connection whether we’re using a Channel Associate Signaling or Common Channel Signaling. And with this we either give up a channel or we don’t, because I mentioned 24 channels in a T1 that’s if you asked me off the top of my head and that’s what I would tell you, but if I chose to dedicate one of those channels as a D-Channel, again just like a BRI circuit that D-channel would be used to send the signaling information down it. The other 23 channels would then be chosen or used for the voice or whatever it is I’m choosing to use those particular channels for, for connectivity. So Common Channel Signaling is going to give you that D-channel. Channel associated signaling means that I’m sending the information and I’m actually kind of robbing a few bits there to send the information down the same channel as my voice traffic or my video traffic is traversing. Now you might say well which one’s better? Well if you have that dedicated D-channel we get a little bit more information from that D-Channel that we can actually leverage in our solution. The Communications Manager actually understands the information coming from that D-Channel, so we may want to really, I don’t want to say waste it, but we may want to dedicate one of our channels of our T1 to be that D-Channel because we can do things above and beyond what we can if we have it tied into the same channel as the phone call.
Converged Voice Networks
So ultimately the goal here is to have a converged voice network. Here are some of the components and topologies you might work with in a Voice over IP solution that has multiple sites, you can see there's a headquarter site, there’s a branch site.
At the headquarters we’ve got Unity Connection for voicemail. We have a dedicated presence server so we can see the status of everybody, we’ve also got the Communications Manager solution out there with some IP phones and then we’ve got a gateway. You see that gateway that says SIP and it connects out to the VoIP provider or there’s also a connection out to the WAN. This is how we might connect out to branch locations. Let’s say we’re supporting this branch location out on the left hand side where we see the Communications Manager Express. We have an IP communicator and of course a switch that all of our devices plug into. So, the branch location and the headquarters location each independently have their own Voice over IP solution, but we’re able to use and leverage the WAN or maybe a VoIP service provider to actually send traffic using RTP between these two locations and that’s one example of a converged network. We’re going to see many in the course of your studies and many in this course.
VoIP Provider Connect
In order for us to connect out to let’s say an IP telephony service provider for example, we would need a special operating system and that special operating system is called the Cisco Unified Border Element. We call it CUBE for short, and what this allows us to do is connect a Voice over IP connection to another Voice over IP connection. In this example we’re talking about service provider, so if I want to connect my network to an IP telephony service provider, I would need to use this Cisco Unified Border Element to interconnect these two networks, and basically what happens here is the CUBE helps us with a demarcation point between the two entities, so between myself and the service provider we can have security, NAT - I may not want my private IP addresses exposed and then we might even want billing to take place. I’m thinking the service provider would probably like to bill me and so with this CUBE we can provide all of this demarcation so that billing can take place, security is in place and we also have two options of how the media or the RTP stream flows through this device. We can either flow through it or we can flow around it, so in other words I could go directly to the person that I am speaking with once the call has setup. So, the call setup takes place through the CUBE, once this call has setup, do we want the RTP stream to go through that or do we have another route, we rather see that RTP stream take to reach the destination and that would be better solution. So it's really just depending upon how you want to set this up. Now with the CUBE environment there's only two Gateway Call Control Protocols that are supported - SIP and H.323, now we can go between H.323 and SIP and vice-versa or we can do SIP-to-SIP or H.323-to-H.323 that part doesn't matter, but those are the only two protocols, Gateway Call Control Protocols that we can select from.