SIP Training

Let’s see if SIP Essentials is a good fit for your training requirements. In this five day class will take a look at the SIP protocol. We will take a detailed look at the Grand picture all the components how they all fit together there’s lots of labs, lots or graphics to make the picture clear. Each chapter will follow this format: We will begin with the vocabulary that will use this vocabulary is probably not necessarily part of your vocabulary so we’ll make sure that we’re not using any terms that you would not be familiar. We also define them straight up at the beginning of each chapter. There’s lots of pictures and diagrams that will describe the concepts were not big in the text lives like this one we actually like to use diagrams of things We always focus on the big picture before we do that deep dive into the protocol we want to make sure you understand how the protocols going to fit into your environment into any environment for that matter. There’s lots of labs and the labs are directly applied to the topics that we are covering furthermore using all freeware so you can replicate these labs at home or at your lab at work. You can also use them to test interoperability in all kinds of possibilities as you may decide when you choose to actually put that in production, at least for testing.

We always finish each section with review questions. now chapter one basically focuses on everything we cover all the topics in this one chapter that the whole course for ultimately provide we want to make sure that you possess the Grand picture about all the components are going to fit together this is the time to make sure that if you’re comfortable with the voice side but the data side is new to you we make sure that that’s understood if telecommunications a somewhat foreign to you do you have a strong data com background we’re gonna speak both your languages in this chapter and make sure there’s a understanding upheld Voice over IP functions, particularly under the control of Session Initiation Protocol (SIP) our emphasis is always on concept proper use a vocabulary we identify all the boxes when we’re finished you’re going to clearly understand the big picture Let me show you an example of one of those slides that we actually see in the first chapter.

Now, this is one of many that we use to describe interoperability this is an example of a SIP call setup that’s interoperating into the public switched telephone network. Let me explain this one just like I would if we were in class. Our “A” party which is originating from 717 566 4428 is calling 215 555 1978 Ultimately, the call setup is going to be hopping across this path, where this portion, at least that far, is going to be public switched telephone network.

The remainder up the signaling path is going to be SIP. We’ll talk about the interoperation right here. This is where the magic is going to occur so our originating party goes off hook, we have thirty milliamps of current flowing and our user dials this phone number you see over here on the right so after that dot the digit 9 215 555 1978 is dialed, a setup messages sent from the PBX to the end office.

Now the setup message contains the calling party’s number, the called party’s number and also the channel that the caller is on. Now that the end office knows this information it must make an autonomous decision based on this information the decision that it makes is part of its routing table known as the dial plan it has its owned.

I’ll play at it did not learn the DOL planned from telephone switches there adjacent to it like you would in an IP network for OSPF or up he IGRP some other interior gateway routing protocol would have told that this is not the case in this particular case a human being has to tell this which how to route calls and the entry then made as a static routing table in this particular case.

Let’s say that are and office chooses this as the best path that it has two hundred in a machine trucks choose from let’s assume that none of those trucks are busy right now well our and office was provisioned to know that signaling system 7 point code 3.5 point no.

I’m is how to talk to our next tandem switch so an initial address messages sent the initial address message specified is going to destination 3.5 point nine from the originating point code 3.5 point to a signal transfer point which looks more like a router really that telephone device simply responded to the destination point good 3.5.9 and relayed initial address message to our tandem switch because it too is configured with a static routing table but this time the static routing tables for point codes.

It’s not for telephone numbers so now are called telephone switch the third one in a row have now knows the originating party’s number the terminating party’s number at also does the call is on channel identification code 19, it must make a routing decision based on its doll play at and its decision is to send that initial address message to our next telephone switch well to our 10 a.m. switch it thinks the next telephone switch is a PSTN or public switched telephone network.

TVM switch and using signaling system 7 the message is sent to 3.5.3 and now our first switch the begins the voice over IP network knows the originating party’s number the terminating party’s number and it knows the calls on channel identification code 22 now it’s time to talk about these little termination points this little termination points are known as a termination specifically these are physical terminations and their patch together with a a logical or a digital Pat scored so that these two terminations interconnect in a context so.

We’ll have a context-free here they’ll be a context in the PBX and in fact right now we have three and a half of them but we begin these Sep networking now the Session Initiation Protocol invite is set first from our tandems which is really what this red this yellow switches to a back to back user-agent: this would be a feature server that happens to know the whereabouts a bar called number because prior to this this end-user registered so the IP address or the demesne over destination is known by the future server the feature server simply relays the SIP invite to where the registration came from now a session border controllers gonna make modifications to the invite making.

It appears though the invite originated from the outside of the session border controller what you should know is that a session border controller has two interfaces one facing in which usually has a private address one facing out which has a public address so when the invite arrives at the session border controller the session border controller completely rewrites the invite making it appears though it’s brand new popping out the outside interface affectively deceiving the called party into thinking that the call actually originated from the session border controller and this is exactly the way we want it this allows the session border controller to hide the rest to the network former outside called party so we would have been untrusted in a trusted side above the session border controller now there’s another thing you should know about the SIP invite which originated here in ultimately relays to the destination.

The SIP invite contains session description protocol now session description protocol is describing a termination a special kind of termination known as an ephemeral termination this is where RTP traffic is going to originate and terminate for this phone call this termination again described by session description protocol must be attached all this information that’s contained in STP must be attached to the SIP invite so when the SIP invite effectively.

Backpacking session description protocol passes through the session border controller the session border controller rewrites session description protocol making it appear as though the ephemeral termination is on the outside a bit self this is gonna cause are called party to send voice to the border controller the called party will never send voice into the sauce which which has an embedded media gateway because it does not know about that in the response to the SIP invite the called you a response ultimately with a sip 200.

This indicates the call was answered and most likely this contains session description protocol I say most likely because there could have been a 183 response prior to this that would contain session description protocol we don’t know that right now so let’s just skip to weave answer the call incessant description protocol describing this and is passed back well session description protocol describing this and his backpack onto the 200 response.

Which affectively back tracks the path that the SIP invite has taken so ultimately the response makes it all the way back to the sauce was with embedded media gateway this is going to permit Voice to travel from the media gateway to the border controller from border controller to destinations so we actually have two separate ephemeral pathways from the media gateway to the inside of the border controller in from the outside at the border controller to the call destination the border control will patch these to a federal terminations inside if itself forming a context which was created completely under the session border controllers control now in all of this please understand the session border controller is forcing it self into this particular call to make sure that it hides the inside from the outside and also the outside from the inside by making it the comment RTP relay where RTP is the voice not the sickling.

It also is a Session Initiation Protocol(SIP) back to back user-agent: this is a special kind of a SIP device which has complete control not only routing but originating terminating and any other aspect of the call which it wants to modifying in any way it’s certainly can so we would look at the control plane as from Saul switch to a back to back user agent from back to back the user agent to border controller on the inside from border controller on the outside to are called USA the media path is going to be coming for on the outside to border controller on the inside to our originating media gateway the the sauce which in this case is not it interested one bit in hearing what is being said it only wants to control the set-up and tear-down the call so now that the calls actually in place we have this blue path if you will it’s all the way back to the originating party and the control path.

which I’ll remind you it use in the red color is the ISDN cue 931 the signaling system 7 messages the SIT messages that you see was all part of the control plane in this particular illustration we have a clear separation between the control plane and this blue stuff or the bearer the result would be that voice packets start to form just like this so we see samples coming in at eight thousand samples per second enough samples are captured until we have it adequate amount normally 20 milliseconds is the favored him out so we capture 20 milliseconds a sound right there and then pass it onward towards the destination this particular Flower Show Inc also a information showing the voice going in this direction we’re not showing the boys going in the other direction for simplicity but believe me it would be occurring so there you have an example house ep RTP session description protocol inner operate with the public switched telephone network.

We have other examples just like this one to show other call scenarios in an attempt to make it crystal clear the big picture house at in or operates within the public switched telephone network will do the same was presence will do instant message and just about everything else that it was controlling these days even video in section 2 we’re going to cover the SIP architecture in SIP architecture will make sure you understand what I use a region is what a proxy is what redirect forking back to back user agents and back to back user agents not only for feature servers and routing but also back to back user agents for security other was in a session border controller.

We’ll talk a little bit about the IP Multimedia Subsystem for the IMS just so you know that’s it has a huge role to play in that and it’s a good way to understand sip architecture we’ll talk about RTP RTCP these are the media channel and the media cue OS reports session description protocol was discussed very lightly in here enough to understand the architecture in more depth we will cover the set methods such as registered invite back refer by cancel and so on make it very clear what each one of these particular set methods or sip requests do we’re going to cover the SIP responses.

Now these the numeric responses the comeback as a result of a request if we send an invite we expect to get back some sort of provisional response by to 18 either 100 or 1.83% and that ultimately a final response that would be a response greater than or equal to 200 during this particular section we’re also gonna talk about via route the record drought this continues on into the next section because Sep managers the sip dialogue in a very clever way the actual decision as to how to setup the city dialogue is done in this set invite all subsequent messages take advantage of this where sip is effectively follows a rule over out once switch many in the next section we’re going to talk about sip your eyes you have to be able to read them the section doesn’t take long may be about 15 or 20 minutes but.

We’ll talk about first of all the definition of who you are I versus URL and then specifically how to read a SIP you are I the colon the semicolon the at sign question marks all those special characters how to read sopranos both for two main parameter as well as a user parameter will apply this not only to a certain invite but the president’s instant message and registrations as well if you ask us to cover certain vendors like you might be interested in how was linked to enter house Avaya doing their house Cisco doing it its all good to us we do not study a protocol like Session Initiation Protocol.

Just for one particular vendor we work with all of them and we like all other they’re all good to us in our next section where take a close look at session description protocol now STP your call describes the ephemeral termination it takes to system description protocols in order to set up a SIP call STP can only describe one and therefore we need STP from the a party we will need STP from the beach party the exchange of session description protocol is done using a process known as offer answer which is part of the median negotiation you need to clearly understand how that works not only because that will greatly help configure it but if you do have to troubleshoot typically it’s not set that’s the problem it’s S and description protocol there are seven key feels that you need to understand in STP just about all problems solved if you know them that we’re going to cover them in great detail the next section talks about the NS well ins in Sep when we are transmitting a sip message to aid in Maine which is not understood by a proxy because you see what our proxy perceives a mess it’s like a sip invite the first thing it does is it looks for the at sign now to the right at the at sign is going to be a dummy what if the demesne does not belong to the SIP proxy well in the previous version of Sep RFC 2543 it would simply discarded this was no mister crowded it would throw the invite away wouldn’t process it but in the new version RFC 3261 which is what some 12 years old now um it’s a different story there is a interesting DNS process nap their servant a record queries that will allow a SIP proxy to relay a message to the proper destination.

It’s extremely well decide you’ll enjoy this chapter its whatever students miss favorite in the next section we talk about me know well this is an extension of the DNS then it’s the perfect place in the course because once we understand the DNS we can study how to take a phone number effectively ride it backwards put in the 16 four-dot arpa suffix on the end do a DNS inquiry which will make sense after studying the previous chapter the results will be the destination to make this means you could take any phone number converted into a domain and and relay this it passes to the appropriate destination very cool and we cover the whole mechanism in enough predominately in the enterprise configuration phones is done with the HCP Noel a SIP has its own DHCP options.

We will cover them this chapter is going to last a whole 10 minutes because no offenders actually following it I but we will cover other vendors and their mechanisms on how to configure and and point to No outbound proxy and proxy for registration in our next section will talk about inter operating Sep with the legacy public switched telephone network so we’ll look at how will this work into work you sick environment how does this work into signaling system 7 or public switched telephone network how much just in our operating indoor PRI what does this do to the way that supports and it turns out that there are some serious consideration sip to solve the problem or I should say people now understand it’s it’s always could solve the problem but we’re starting to deploy it properly these days you need to understand what a 183 is we need to define what call progress means and ultimately out tell you right now it means that the progress tones are in the media path which means you actually have to listen with your years to determine what’s going on because the signaling protocols not going to tell you in this section will also study sip tea for those view more involved in the public switched telephone network side this is a big deal to you will cover that here in the next section we talk about RTP this is the media path so RTP is force over IP RTP is how we encapsulated Voice make sure that we timestamp it at a identifiers like a microphone identify that identifies who’s talking there’s other fields that are involved so the when the RTP encapsulated Voice arrives the destination it can be played back as if the person we’re standing there right in front of you in real time.

We create the illusion that over and a synchronous network we actually provide synchronous information RTP has the magic told to do this it’s known as a timestamp the jitter in can occur because at this point explain that mechanism now it might be important that you track how well our TP is behaving there’s a protocol for that it’s known as RTP control protocol RTCP now RTCP if you choose to run it will add another five percent love overhead to the voice call but as a result you all know at the end of the call how all those performance plus will earn profit loss latency and jitter from this and in this chapter if you don’t know what that is we’ll explain that as well in our next section we’re going to take a look at DTMF this is an extension actually have the previous chapter the subject was simply too broad to pack it into the RTP chapter when we dialed digits we’re actually creating two-tone simultaneously this is dual tone multi-frequency there are problems using DTMF across a network that is using some compression.

I wear them like speaks or G 729 if you using protocol if you using compression like that the DTMF tones are damaged you should learn signal to noise ratio twist amplitude and other issues so that are causing the problem so that when you dial into voicemail are you calling auto attendant and no matter how hard you da how long you press the buttons you can at get the other and to respond all explain exactly why that’s the case the solution is while you could switch back to G 711 you could use RFC 2833 or sip info RFC 2833 that is sort of an example love rather than actually said the recording with the tone sort of like we send music so it said were playing a half note a ver si and a half note other event be play that particular of third cord at the other end.

That’s an example of RFC 2833 now in the SIP info message you say well we’re we’re playing a four digit where I could tell you what the duration other this I if this works it has advantages there’s a lot more will cover that stuff in this chapter in our next section which all too often times as optional students either care very strongly or they don’t care at all regarding fax and elect will address this in a number of ways if we have two or three students that feel very strongly about this chapter we might just push it all the way to the back and they can hang around to the end which course and will cover that if we’re all interested or maybe you want some degree of it how will cover it we can go to whatever depth you want to in fact handling will lease cover T-thirty and t38 an RTP relay use in G 711 their next section we’re gonna cover presence so presence is a combination of instant message with your login or logged out its all based on XML or text or HTML of some kind markup for making our text messages look really cool and add the smiley faces and all that sort of thing oftentimes.

I note that when students take this particular sections are thinking that presence is like sky that or something like that that’s going to its gonna keep track above %uh what we’re all doing to a that’s not really the way the protocol works although certainly it could thus nothing stopping you from doing it that way it’s more of a relay service so that if your status changes you would include that in a sec notify republish message relay it onto a presence server which relays anybody who cares than they would have described that they care using a set subscribe so explain exactly how that works it’s a fun chapter it doesn’t take very long to understand presence once you understand but a SIP dialog is believe me its kits its probably the easiest chapter in the course once you understand how supports in our next section we talk about set timer sip is designed to work over you DP therefore if isset message for some reason is dropped in the IP network a we need to retransmit well the other way to handle that is we need some sort of it no argument coming back so we need to know how long we wait around before some message comes back before we try again and if we do try again we still don’t get a response back well then we retransmit but then how many times a week retransmit before just completely give up it turns out it depends on the set method this section is going to cover that also in this section.

We’re going to talk about the session expires timer this one is primarily have interest to mobile carriers these are people that are in the cellular business it’s possibly do a battery pull drive through a tunnel go over a mountain or something like that and you no longer can keep track of the guys there to just go on abut the mobile network would think that they’re still on a call the call would stay in the held up so there needs to be a way to establish a heartbeat to make sure the persons there that’s the session expires timer will take a close look at that we have a great section onset security 06 sip security section is going to cover all the details regarding t last sax PK I a symmetric key a and asymmetric he we’re going to get into our essay will talk about certif certificates will talk about fake certificates and how you should deploy a PK I infrastructure if that’s your plan we think you should understand on a lot more the mechanism than perhaps a good bargain for I think it’s more than just copying route keys and things like that you should understand the mechanism and ultimately what you should understand is after all the work is it really secure anyhow now we’re white hats were not black cats I suppose.

We could be the bad guys would want to but that’s not our plan whatsoever so we’re going to show you how to keep the black cats at bay we don’t teach hacking that’s not something that all the three deaths in our next section a talk about NAT traversal its have the perfect time in the course cuz at this particular time we have covered system description protocol we understand RTP we’ve taken a look at Sep it turns out that in order to avoid the classic one way voice syndrome what’s now that’s cause we need to put some sort of mechanism in place to make sure that voice gets through the net well there’s several different mechanisms their stun just turned we’re gonna cover ice now these mechanisms actually extend beyond set up if you’re studying the new html5 stuff particularly on media from browser to browser they’re using the same techniques and we try to do that in all of our courses.

We just like technology sometimes just for its sake even the especially when technology bleeds into many other environments we’re going to make sure that you understand that’s the case in this particular case not traverse all axle is impacted by a lot of different technologies if you understand it you’re not going to have one way voice at least not from thats in the next section we’re going to cover set P now this’ll be our last section it’s a great chapter to end up with this is a big hit for quality assurance people for testers you think people like that because sit p is the perfect tool to test yourself environment so maybe your saddled with testing to see how many hundreds of thousands of calls per second your SIP proxy can process maybe you’re trying to replicate the problem a to to see if you come up with some sort of solution maybe you just trying to understand the set protocol anyone others possibilities sipes a perfect day.

It was given to us by Hewlett Packard they did a great job with that their scepter there sa subside actors other tools which we will use in here but we likes it P the most because it’s xml-based its easy to modify other students pick it up quickly and of course it’s free so you can use it in your environment for testing finally we have some really nice labs they’re listed in the bottom of the page if you just look great down there we have them all listed shown here are few if you the highlights you gonna set up a sip lab we use into Maley oh and they will use asterisks we use camellia because we need a SIP proxy to illustrate the difference between a set property and a back to back user-agent: most to the course begins using camilia which infused ever use this you like this is not an easy proxy to get operating but it’s extremely powerful its carrier-grade it measures capability in the hundreds of thousands of calls per second we’ve tested it were impressed with this product it’s just not for the average bear you need to understand the protocol well to use this but i promisee as a result of this class you won’t be afraid of Camellia.

I’ll and you may even decide to use it at certain places in your network we use Wireshark that will be our primary tool for decoding will show you the washer trips in order to decode Voice SDP we’ll take a look at sip we’ll take a look at how to do flow graphs and and to offer the services graphs that we can weed out certain sent messages will show you which particular fields make sense to filter on finally we’ll take a look at: shark and how you can use T shark to do so probes all around your network you may decide that a freeware solution might be the right one to manage sit on your network at the near the end the class so it’s imperative that we start to play with back to back user agents will put to mail you aside more breakout asterisk will use assets for a number of feature servers but per feature server applications but primarily are interest with Asterisk is just to take a look at how about back user-agent: behaves at this stage all kinds of things can happen during certain private engagements we will get people that say what can you just show us this with a fire can you show it with Cisco maybe you can show with this might I’ll switch that we have four we have no problem with that we will make a dinner operate so fast to make your head spin this is what we do for a living arm in fact one of the courses that we put together the %uh via SIP trunking class predominately because we sell that’s a product where infusion of our understanding would really help it you might want to take a look at that one as well well there you have it that’s the SIP essentials class in a nutshell hope to see you in a class A I’m Stewart Cesar soul of

As found on Youtube

Leave a Reply

Your email address will not be published. Required fields are marked *