let's see if sip essentials is a goodfit for your training requirements. in this five day class will take a lookat the sip protocol. we will take a detailed look at the grandpicture all the components how they all fittogether there's lots of labs, lots or graphics to make the pictureclear. each chapter will follow this format: we will begin with the vocabulary thatwill use this vocabulary is probably not necessarily part of yourvocabulary so we'll make sure that we're not using any terms that you would notbe familiar.
we also define them straight up at thebeginning of each chapter. there's lots of pictures and diagramsthat will describe the concepts were not big in the text lives like this one weactually like to use diagrams of things we always focus on the big picturebefore we do that deep dive into the protocol we want to make sure you understand howthe protocols going to fit into your environment into anyenvironment for that matter. there's lots of labs and the labs aredirectly applied to the topics that we are covering
furthermore using all freeware so youcan replicate these labs at home or at your lab at work.you can also use them to test interoperability in all kinds ofpossibilities as you may decide when you choose to actually put that inproduction, at least for testing. we always finish each section withreview questions. now chapter one basically focuses on everything we cover all the topics in this one chapter that the wholecourse for ultimately provide we want to make sure that you possessthe grand picture
about all the components are going tofit together this is the time to make sure that if you're comfortable with thevoice side but the data side is new to you we makesure that that's understood if telecommunications a somewhat foreignto you do you have a strong data com background we're gonna speak both your languages in this chapter and make sure there's aunderstanding upheld voice over ip functions,particularly under the control of session initiationprotocol (sip)
our emphasis is always on concept proper use a vocabulary we identify allthe boxes when we're finished you're going toclearly understand the big picture let me show you an example of one ofthose slides that we actually see in the first chapter. now, this is one ofmany that we use to describe interoperability this is an example of a sip call setupthat's interoperating into the public switchedtelephone network. let me explain this one just like iwould if we were in class.
our "a" party which is originating from717 566 4428 is calling 215 555 1978 ultimately, the call setup is going tobe hopping across this path, where this portion, at least that far, is going to be publicswitched telephone network. the remainder up the signaling path is going to be sip. we'll talk about theinteroperation right here. this is where the magic is going to occur so our originating party goes off hook, wehave thirty milliamps of current
flowing and our user dials this phone number you seeover 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 messagecontains 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 decisionbased on this information the decision that it makes is part of itsrouting table known as the dial plan it has its owned i'll play at it did notlearn the dol planned from telephone switches there adjacent to itlike you would in an ip network for ospf or up he igrp some other interiorgateway routing protocol would have told that this is not the case in thisparticular case a human being has to tell this which how to route calls and the entry thenmade as a static routing table in this particular case let's say thatare and office chooses this as the best
path that it has two hundred in a machine trucks choose from let'sassume that none of those trucks are busy right now well our and office was provisioned toknow that signaling system 7 point code 3.5point no i'm is how to talk to our next tandem switch so an initial address messages sent theinitial address message specified is going to destination 3.5 point nine from the originating point code 3.5point to
a signal transfer point which looks morelike a router really that telephone device simplyresponded to the destination point good 3.5.9 and relayed initial address message toour tandem switch because it too is configured with astatic routing table but this time the static routing tablesfor point codes it's not for telephone numbers so now are called telephone switch thethird one in a row have now knows the originating party'snumber the terminating party's number
at also does the call is on channelidentification code 19, it must make a routing decision basedon its doll play at and its decision is tosend that initial address message to our next telephone switch well to our 10 a.m.switch it thinks the next telephone switch is apstn or public switched telephone network tvmswitch and using signaling system 7 the message is sent to 3.5.3and now
our first switch the begins the voiceover ip network knows the originating party's number theterminating party's number and it knows the calls on channel identification code22 now it's time to talk about these littletermination points this little termination points are knownas a termination specifically these are physicalterminations and their patch together with a alogical or a digital pat scored so that these two terminationsinterconnect in a context so we'll have acontext-free here
they'll be a context in the pbx and infact right now we have three and a half ofthem but we begin these sep networking now the sessioninitiation protocol invite is set first from our tandems which is really what this redthis yellow switches to a back to back user-agent: this wouldbe a feature server that happens to know the whereabouts abar called number because prior to this this end-user registered so the ipaddress or the demesne over destination is known by the futureserver
the feature server simply relays the sipinvite to where the registration came from nowa session border controllers gonna make modifications to the invite making it appears though the inviteoriginated from the outside of the session border controller what you should know is that a sessionborder controller has two interfaces one facing in which usuallyhas a private address one facing out which has a public address so when the invite
arrives at the session border controllerthe session border controller completely rewrites the invite making it appearsthough it's brand new popping out the outside interfaceaffectively deceiving the called party into thinking that the call actuallyoriginated from the session border controller and this is exactly the way we want itthis allows the session border controller to hide the rest to thenetwork former outside called party so we wouldhave been untrusted in a trusted side above the sessionborder controller
now there's another thing you shouldknow about the sip invite which originated here inultimately relays to the destination the sip invite contains sessiondescription protocol now session descriptionprotocol is describing a termination a special kind oftermination known as an ephemeral termination thisis where rtp traffic is going to originate andterminate for this phone call this termination again described by session description
protocol must be attached all this information that's contained instp must be attached to the sip invite sowhen the sip invite effectively backpacking sessiondescription protocol passes through the session bordercontroller the session border controller rewrites session description protocol making itappear as though the ephemeral termination is on theoutside a bit self this is gonna cause are called party
to send voice to the border controllerthe called party will never send voice into the sauce which which has anembedded media gateway because it does not know about that in the response to the sip invite thecalled you a response ultimately with a sip 200 this indicates the call was answered andmost likely this contains session description protocol i say most likely because there couldhave been a 183 response prior to this that would contain sessiondescription protocol
we don't know that right now so let'sjust skip to weave answer the call incessant descriptionprotocol describing this and is passed back well sessiondescription protocol describing this and his backpack onto the 200 response which affectivelyback tracks the path that the sip invite has taken so ultimately theresponse makes it all the way back to the sauce was with embedded mediagateway this is going to permit voice to travelfrom the media gateway to the border from border controller
to destinations so we actually have twoseparate ephemeral pathways from the mediagateway to the inside of the border in from the outside at the bordercontroller to the call destination the border control will patch these to afederal terminations inside if itself forming a context which was createdcompletely under the session border controllers control now in all of this please understand thesession border controller is forcing it self into this particular call to make sure that ithides the inside from the outside and
also the outside from the inside by making it the comment rtp relay wherer tp is the voice not the sickling and it also is a session initiationprotocol back to back user-agent: this is a special kind of a sip devicewhich has complete control not only routing but originatingterminating and any other aspect of the call whichit wants to modifying in any way it's certainly can so we would look at the control plane asfrom saul switch to a back to back user agent from back to back the user agentto
border controller on the inside fromborder controller on the outside to are called usa the media path isgoing to be coming for on the outside to border controller on the inside toour originating media gateway the the sauce which in this case is notit interested one bit in hearing what is being said it onlywants to control the set-up and tear-down the call so now that the calls actuallyin place we have this blue path if you will it's all the way back to theoriginating party and the control path which i'll remindyou it
use in the red color is the isdn cue 931the signaling system 7 messages the sit messages that you see was allpart of the control plane in this particular illustration we havea clear separation between the control plane and this blue stuff or the bearer the result would be that voice packetsstart to form just like this so we see samples comingin at eight thousand samples per second enough samples are captured until wehave it adequate amount normally 20 milliseconds is the favored him out
so we capture 20 milliseconds a soundright there and then pass it onward towards thedestination this particular flower show inc also a information showing the voice going inthis direction we're not showing the boys going in theother direction for simplicity but believe me it would be occurring so there you have an example house eprtp session description protocol inneroperate with the public switched telephonenetwork we have other examples just like
this one to show other call scenarios in anattempt to make it crystal clear the big picture house at in or operates within the public switched telephonenetwork will do the same was presence will do instant message and just about everything else that itwas controlling these days even video in section 2 we're going tocover the sip architecture in sip architecture will make sure youunderstand what i use a region is what a proxy is what redirect forking
back to back user agents and back toback user agents not only for feature servers and routing but also back toback user agents for security other was in a session border controllerwe'll talk a little bit about the ip multimedia subsystem for the ims just so you know that's it has a hugerole to play in that and it's a good way to understand siparchitecture we'll talk about rtp rtcp these are themedia channel and the media cue os reports session descriptionprotocol was discussed very lightly in here enough to understand thearchitecture
in more depth we will cover the setmethods such as registered invite back refer by cancel and so onmake it very clear what each one of these particular setmethods or sip requests do we're going to cover the sipresponses now these the numeric responses the comeback as a result of a request if we send aninvite we expect to get back some sort of provisional response by to 18 either 100 or 1.83% and that ultimately a final responsethat would be a response greater than or
equal to 200 during this particular section we'realso gonna talk about via route the record drought this continueson into the next section because sep managers the sip dialogue in a very clever way the actual decisionas to how to setup the city dialogue is done in thisset invite all subsequent messages take advantageof this where sip is effectively follows a ruleover out once switch many in the next section we'regoing to talk about sip your eyes you
have to be able to read them the section doesn't take long may beabout 15 or 20 minutes but we'll talk about first of all the definition of who youare i versus url and then specifically how to read a sipyou are i the colon the semicolon the at signquestion marks all those special characters how to readsopranos both for two main parameter as well as a user parameter will apply this not only to a certaininvite but the president's
instant message and registrations aswell if you ask us to cover certain vendorslike you might be interested in how was linked to enter house avaya doing theirhouse cisco doing it its all good to us we do not study aprotocol like session initiation protocol just for one particular vendor we work withall of them and we like all other they're all good to us in ournext section where take a close look at session description protocol now stp your call describes theephemeral termination it takes to
system description protocols in order toset up a sip call stp can only describe one and therefore we need stp from the aparty we will need stp from the beach party the exchange of session descriptionprotocol is done using a process known as offer answer which is part of the mediannegotiation you need to clearly understand how thatworks not only because that will greatly helpconfigure it but if you do have to
troubleshoot typically it's not set that's theproblem it's s and description protocol there are seven key feels that you needto understand in stp just about all problems solved if youknow them that we're going to cover them in greatdetail the next section talks about the ns well ins in sep when we are transmitting a sip message to aid in maine which is notunderstood by a proxy because you see what our proxy perceivesa mess it's like a sip
invite the first thing it does is itlooks for the at sign now to the right at the at sign is goingto be a dummy what if the demesne does not belong tothe sip proxy well in the previous version of sep rfc2543 it would simply discarded this was no mister crowded it wouldthrow the invite away wouldn't process it but in the newversion rfc 3261 which is what some 12 years oldnow um it's a different story there is a interesting dns process naptheir servant a record queries
that will allow a sip proxy to relay amessage to the proper destination it's extremely well decide you'll enjoythis chapter its whatever students miss favorite in the next section we talk about meknow well this is an extension of the dns then it's the perfect place in thecourse because once we understand the dns we can study how to take a phone numbereffectively ride it backwards put in the 16 four-dot arpa suffix onthe end do a dns inquiry which will make senseafter studying the previous chapter
the results will be the destination tomake this means you could take any phonenumber converted into a domain and and relaythis it passes to the appropriate destination very cool and we cover the whole mechanism inenough predominately in the enterprise configuration phones is done with thehcp noel a sip has its own dhcp options we will cover them this chapter is goingto last a whole 10 minutes because no offenders actually following it i butwe will cover
other vendors and their mechanisms onhow to configure and and point to no outboundproxy and proxy for registration in our next section will talk aboutinter operating sep with the legacy public switchedtelephone network so we'll look at how will this work intowork you sick environment how does this work into signaling system7 or public switched telephone network how much just in our operating indoorpri what does this do to the way thatsupports and it turns out that there are someserious consideration
sip to solve the problem or i should saypeople now understand it's it's always could solve the problem but we'restarting to deploy it properly these days you need to understand what a 183 is weneed to define what call progress means and ultimately out tell you right now itmeans that the progress tones are in the media pathwhich means you actually have to listen with your years to determine what's going onbecause the signaling protocols not going to tell you
in this section will also study sip teafor those view more involved in the public switched telephone network sidethis is a big deal to you will cover that here in the next sectionwe talk about rtp this is the media path so rtp is force over ip rtp is how we encapsulated voice make surethat we timestamp it at a identifiers like a microphoneidentify that identifies who's talking there's other fields that are involved so thewhen the rtp
encapsulated voice arrives thedestination it can be played back as if the personwe're standing there right in front of you in real time we create the illusion thatover and a synchronous network we actuallyprovide synchronous information rtp has the magic told to do this it'sknown as a timestamp the jitter in can occur because at thispoint explain that mechanism now it might be important that you trackhow well our tp is behaving there's a protocol for that it's knownas rtp control protocol
rtcp now rtcp if you choose to run it will add anotherfive percent love overhead to the voice call but as aresult you all know at the end of the call howall those performance plus will earn profit losslatency and jitter from this and in this chapter if youdon't know what that is we'll explain that as well in our next section we're going to takea look at dtmf this is an extension actually have theprevious chapter the subject was simply
too broad to pack it into the rtpchapter when we dialed digits we're actuallycreating two-tone simultaneously this is dualtone multi-frequency there are problems using dtmf across a network that is using somecompression i wear them like speaks or g 729 if you using protocol if you using compression likethat the dtmf tones are damaged you shouldlearn signal to noise ratio twist amplitude and other issues
so that are causing the problem so thatwhen you dial into voicemail are you calling autoattendant 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 switchback to g 711 you could use rfc 2833 or sip info rfc 2833 that is sort of anexample love rather than actually said therecording with the tone
sort of like we send music so it saidwere playing a half note a ver si and a half noteother event be play that particular of third cord atthe other end that's an example of rfc 2833 now in the sip info message you say wellwe're we're playing a four digit where i could tell you whatthe duration other this i if this works it has advantagesthere's a lot more will cover that stuff in this chapter in our next section which all too often times as optionalstudents either care
very strongly or they don't care at allregarding fax and elect will address this in a number of ways ifwe have two or three students that feel very strongly about this chapter wemight just push it all the way to the back and they can hang around to the endwhich course and will cover that if we're all interested or maybe youwant some degree of it how will cover it we can go to whateverdepth you want to in fact handling will lease cover t-thirty and t38 an rtprelay use in g 711 their next section
we're gonna cover presence so presenceis a combination of instant message with your login orlogged out its all based on xml or text or html of some kind markup for making ourtext messages look really cool and add the smiley faces and all that sort ofthing oftentimes i note that when studentstake this particular sections are thinking that presence is like sky thator something like that that's going to its gonna keep track above %uh whatwe're all doing to a that's not really the way the protocol works althoughcertainly 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 notifyrepublish message relay it onto a presence server whichrelays anybody who cares than they would havedescribed that they care using a set subscribe so explain exactly how thatworks it's a fun chapter it doesn't take very long to understand presence once youunderstand but a sip dialog is believe
me its kits its probably the easiestchapter in the course once you understand how supports in ournext section we talk about set timer sip is designed to work over you dp therefore if isset message for somereason is dropped in the ip network a we need to retransmit well the other way to handle that is we needsome sort of it no argument coming back so we need to know how long we waitaround before some message comes back before we try again and if we do tryagain we still don't get a response back
well then we retransmit but then howmany times a week retransmit before just completely give up it turns out it depends on the setmethod this section is going to cover that alsoin this section we're going to talk about the session expires timer this one is primarily have interest to mobile carriers these are people thatare in the cellular business it's possibly do a battery pull drivethrough a tunnel go over a mountain or something like that
and you no longer can keep track of theguys there to just go on abut the mobile network would think thatthey're still on a call the call would stay in the held up so there needs to be a way to establisha heartbeat to make sure the persons there that'sthe session expires timer will take a close look at that we have a great section onset security06 sip security section is going to cover allthe details regarding t last sax pk i
a symmetric key a and asymmetric he we're going to get into our essay willtalk about certif certificates will talk about fakecertificates and how you should deploy a pk i infrastructure if that's your plan wethink you should understand on a lot more the mechanismthan perhaps a good bargain for i think it's more than just copyingroute keys and things like that you should understand the mechanism andultimately what you should understand is after all the work is it really secure
anyhow now we're white hats were notblack cats i suppose we could be the bad guys wouldwant to but that's not our plan whatsoever so we're going toshow you how to keep the black cats at bay wedon't teach hacking that's not something that all the threedeaths in our next section a talk about nat traversal its have theperfect time in the course cuz at this particular time we have covered system description protocol weunderstand rtp we've taken a look at sep it turns outthat in
order to avoid the classic one way voicesyndrome what's now that's cause we need to put some sort of mechanism in place to make sure that voice getsthrough the net well there's several differentmechanisms their stun just turned we're gonna cover ice now thesemechanisms actually extend beyond set up if you're studying the new html5 stuff particularly on media from browser tobrowser they're using the same techniques and we try to do that in all of ourcourses
we just like technology sometimes justfor its sake even the especially when technologybleeds into many other environments we're going to makesure that you understand that's the case in this particular case not traverse allaxle is impacted by a lot of different technologies ifyou understand it you're not going to have one way voice at least not from thats in the nextsection we're going to cover set p now this'llbe our last section it's a great chapter to end up with thisis a big hit for quality assurance
people for testers you think people likethat because sit p is the perfect tool to test yourself environment so maybeyour saddled with testing to see how many hundreds of thousands of calls persecond your sip proxy can process maybe you're trying to replicate theproblem a to to see if you come up with some sortof solution maybe you just trying to understand the set protocol anyone others possibilities sipes aperfect day it was given to us by hewlett packardthey did a great job with that
their scepter there sa subside actorsother tools which we will use in here but we likes it p the most because it'sxml-based its easy to modify other students pick it up quickly and ofcourse it's free so you can use it in your environmentfor testing finally we have some really nice labsthey're listed in the bottom of the page if you just look great down there we have them all listed shown here arefew if you the highlights you gonna set up a sip lab we use intomaley oh and they will use asterisks we use camellia because we need a sipproxy
to illustrate the difference between aset property and a back to back user-agent: most to the course begins using camiliawhich infused ever use this you like this isnot an easy proxy to get operating but it's extremely powerful itscarrier-grade it measures capability in the hundredsof thousands of calls per second we've tested it were impressed with thisproduct it's just not for the average bear youneed to understand the protocol well to use this
but i promisee as a result of this classyou won't be afraid of camellia i'll and you may even decide to use it atcertain places in your network we use wireshark that will be ourprimary tool for decoding will show you the washer trips in orderto decode voice sdp we'll take a look at sip we'lltake a look at how to do flow graphs and and to offerthe services graphs that we can weed out certain sent messageswill show you which particular fields make sense to filter on finally we'll take a lookat: shark
and how you can use t shark to do so probes all around your network you maydecide that a freeware solution might be the right one to manage sit on your network at the near the end the class so it'simperative that we start to play with back to back user agents will put to mail you aside more breakoutasterisk will use assets for a number of featureservers but per feature server applications butprimarily are
interest with asterisk is just to take alook at how about back user-agent: behaves at this stage all kinds of things can happen duringcertain private engagements we will get peoplethat say what can you just show us this with a fire can you show it with cisco maybeyou can show with this might i'll switch that we have four we have no problem with that we willmake a dinner operate so fast to make your head spin this is what we do for aliving
arm in fact one of the courses that weput together the %uh via sip trunking class predominately because we sell that's aproduct where infusion of our understandingwould really help it you might want to take a look at thatone as well well there you have it that's the sip essentials class in anutshell hope to see you in a class a i'm stewartcesar soul of
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