all right so in this video we're going to talk about the call setup and your hear down process now obviously in order to troubleshoot a teleporting network you need to understand the process of has the call being set up and what happens when the call hands on so that you have a better understanding about how to troubleshoot the network and with the detailed understanding about the whole process you will be able to identify when you are analyzing the packet as what's happening during each process such as are we receiving what we expected uh part of our scanning procedures so false setup which initially initiating a call so obviously a user will uh has lifted the handset receive a dial tone now that dial tone is coming from the call manager server or cucm they will dial the party's poll party's number so in this case for example 1000 is dialing 2001. so the intention is to call that particular person right about there these digits will be sent to the call control agent in this case this call manager one at a time depending on the type of phone that you're using we will talk about the digit uh uh this numbering concept later in the course but right now understand that the digit will be sent to the poly controller call control agent cuc amp which could be a voice enabled router or could be a cucm manager now once that part is done your next step is for endpoint discovery the call control agent which is the call manager right here will then need to identify the location of the number that you dial in this case the called party this endpoint could be registered locally to this call manager x or could be registered remotely to call manager y so you need to understand where the call manager needs to understand where is this device located it might be require a routing configuration to route the call to a remote destination so once it identifies that your next step is to permission check so call manager is going to say hey does this guy have the formation the permission check may be performed to confirm the calling party in this case the this particular phone has the right to dial that number so that is one of the thing that will happen before even signal is sent to the remote site so once the permission check is done then comes to bandwidth chat because remember we could confuse call admission control to restrict the bandwidth or your network may not have enough necessarily bandwidth so in this scenario in the during the bandwidth check the call control agent in this case call manager may check to determine if there is enough sufficient bandwidth on the network to even allow the call the bandwidth check is better known as call it machine control so this call admission control for example is going to be part of your uh administration configuration and you want you want to make sure that you have a proper understanding about that now bandwidth check is better known as sorry vendor check can be called mission control or it could be rsvp so providing that these two information has been successful then the call progress tone steps take place in this case once the call party endpoint is ringing this k i mean 2000 phone is ringing then a progress stone has to be sent back to the caller right so this guy know that the call is in progress otherwise people would assume that there is something wrong in the network if the call party is engaged the engage progress stone will be played back which is another word visitor so there are two examples of call progress tone that are available during this process now once the call has been answered once the call party answered the call the progress tones are no longer needed a cdr in this after once you answer the call at that moment the call control agent the cuc amp can be configured to log all call information logging can be done through several different technologies such as syslog server radius server or direct database entry the method used for logging will largely depend on the type of cloud control agent that you're using for example if you are using call manager cuc amp in this case logging can be done directly into the database whereas if you're using cisco router as a call control device such as call manager express or cu cme in that case syslog or radius server can be configured to achieve that now once the call detailed recording take place then we do nega capability negotiation this is where the product negotiation happens the capability negotiation exchange is required in order to find a common supported and requested codec between the two endpoints both party has to agree to the same codec both party has to agree to all the bite size and all the all the other necessary negotiat capabilities if for some reason one of the negotiation has failed if chances are that either the call will fail or you may need an additional device to make that call work think about in a scenario where you and i are speaking different language so let's say you speak english i'm speaking hindi or indian you know a different language obviously we may not uh understand each other unless we either speak the same language in this case english or we need an interpreter so capability negotiation will identify those shortfall and substitute whatever can based on what is configured to either include an interpreter in this case transporter or could we just simply say i can't negotiate so therefore i'm going to send you fast busy signal which will basically indicate that the call has failed now once the call negotiation is done then negotiating the stream the two endpoint in this case the two phone will then negotiate the port that they want to be used to establish a bi-directional connection or real-time transport protocol or known as rtp and as well as rtcp rtp is the protocol that carries the conversation so if i say hello world it's not carry that conversation into that packet but it used a port number udp port so it will have to negotiate from this range what port do you guys want to use to transport the packet and you could at the same time it's going to negotiate an rtcp so rtcp or real-time con real-time transport control protocol is basically we'll use the rtp 4 plus 1 to pair the connection which is used for monitoring the quality of the rtp or monitoring the behavior of the rgb protocol so what rtcp does it monitors the rtp flow and kind of send a feedback to the following party say look i am experiencing problem in the network there's a jitter there's a packet loss something is happening so the quality of the call or rtp flow will be reported back to the column uh calling partner so that in order to do that is going to know which rtp flow to monitor right because there could be 100 of rtp flow well if your rtp flow use if your rtp use port number 16384 let's say 16384 in that case rtcp will be using 16384 plus one so rtp if it is odd number or even number doesn't matter it's always going to use whatever the rtv flow plus one to indicate that that is flow uh monitoring that particular rtp session by simply subtracting that so if you are looking at an rtcp portal rtcp port 16385 that means that is for rtp 16384 so that's an easy way to figure this out now once you're finished talking you're ready to hang up the call you simply then you know hang up so when either calling or call party terminates the call by hanging up the end point a signal is sent to the call manager in this case call control agent to tear down the call and release the resources such as memory and all other stuff closed connection a signaling between the call control agent it will be sent to notify and acknowledge the termination of the call and release the resource then it goes back to cdr cdr will report the call as call being completed bandwidth will be released so that it can be located for all other calls so these are the steps that take place as the call is in progress and it is our job as an engineer to have a good understanding of that so that we know exactly where the call could fail and why the call may have failed so that is an example of the call flow of your uh ip telecontent this is an example of a say a sip called flow now i'm going to walk you through step by step what happened when an initiating party initiates a call so end point such as let's say i am the endpoint i'm gonna pick up the call to call you so i initiate the call i will be known as user agent client or uac and i will be sending an invite message to my user agent server in this case the call manager so the user is called sorry let me rewrite it user is called uac and this guy called manager will be called uas user agent server and i'll be sending an invite message now in this case the user agent server call manager receives the response to the invite message using 100 so it's going to use uh rece uh sorry uh if in step number one initiate the call signal is sent it's gonna receive an acknowledgement with 100 trying messages now the terminating gateway sends a signal to the receive recipient's telephone in this case it's going to send a signal to this one that hey you are receiving a call step number five the recipient user agent server informs a user agent client that about ringing signals with 180 so which is the game this guy's sending a signal back to saying hey call is being taking place so now both sides are receiving the ringing step number six where originating gateway sends the ring back tone to the caller so this guy will send a ring back saying hey the call is in progress right now step number seven the called telephone is taken off the hook so now after all the ringing and everything what not this guy decided to answer the call so he picked up the handset right so the call party is taking off the hook and step number seven step number eight that if the user agent server this particular call manager uas okay issues uh sorry step number eight the user agent server of the recipient the la this party determined that the call parameters are acceptable in all the negotiations and everything it responds positively to the user agent client which is this call manager saying okay everything is okay so it will send an acknowledgement in step number nine okay step number nine originating usc will then issue an acknowledgement so step number nine is call manager sending an acknowledgement saying okay i received your uh information now step number 10 at this point the user agent fine and user agent server have all the information that is required to establish an rtp stream so they're going to negotiate the port number rtcp port number what not between them step number eleven one of the party decided that okay i'm gonna hang up the call it's send uh you send a buy message to the user agent so let's say this guy hang up the call he send a buy message to this guy this guy will send a buy message to this guy or vice versa it doesn't matter the buy message is confirmed by acknowledging 200 okay that yeah i received your buy message we're all good to go have a nice day so it's kind of like i complete the call sip will use other standard-based protocol to provide large set of features based on relatively simple mechanism and we're going to talk about what is that mechanism it's called session discovery protocol ldp session discovery protocol is an ietf based format for describing streaming media initialization parameter in ascii stream sdp is intended intended for describing the media connection oh sorry media communication session for the purpose of session announcement session invitation parameter negotiation all these are part of session discover uh session description uh descript session disc description protocol or sdp sdp does not deliver media itself but it is used for negotiating negotiation between the endpoint of media type media format and all associated properties set up properties set up properties and parameters are often called the session profile sdp is extensible extensible to support new media type and format now this figure will show you there are two sdps example the table explains the parameter what they are used for so if you take a look at the first one example number one is an audio call how do i know well look at the media m is equal media type and it's an audio media right so obviously that's an audio call it also tells you what is the rdp rtp port number and also tells you um various other things like for example what codec is going to use so if you take a look at carefully you'll see rtp four nine one zero nine and this zero that you see right there that's an indication it is z g dot seven eleven if it is other than j zero it could mean something else okay if you take a look at example number two it's also an audio call it was a video i'll show you video it is using the port number plus it has offered three different product g.729 which is 18 g.711 which is zero and g dot seven eleven a law which is uh eight so you got three uh three choices are made g dot seven twenty nine eighteen zero is g dot seven eleven a mulan which is north america and g.711 a law which is the third choice a so by looking at this number you can automatically know what product was offered so it is important that you understand these values so that when you're troubleshooting your sip you're gonna see this stuff so let's go through this item what they basically does uh v stands for version so v equals zero that's the default version then you'll see o o stands for origin origin means username session id version network type address type and address itself it contains multiple information in the origin then you got s session name what is the name of a session it could be your username who knows time t stands for time uh start time and the stop time okay uh c stands for connection data okay connection data means what ip address it was used to establish the connection see here you find the ipaddress and what type of ip protocol is it ipv4 or ipv6 and then you got m stands for media okay it will get media the port the transport and the connection address where it's connecting to uh audio means um the codec will offer zero equal g dot seven eleven new law which is for north america canada u.s and japan japan maybe and you got uh eight g seven eleven a law three is gsm product and eighteen is your g7 g.729 these are your information that can help you troubleshoot or identify the problem as we go through this course so make sure you are familiar with them make sure you can read what that what is supposed to mean because if your job involves in analyzing the trace files you'll see lots of this stuff and it may help you may be required for you to understand those log files by reading through this information to give you an idea what you're dealing with now one other thing about sdp is that how am i going to offer these as sdp to you if i'm calling you how am i going to offer this to you right so there are two ways to exchange the sdp output and answer messages now these methods are commonly known as delay offer and early offer and it supports for both method by user agent and clan server is a mandatory so it's important that both device supports that in simplest term the initial the initial safe invite that if i'm calling you my sip invite to you is the initial the initial sip invite that is sent with the sdp in the message body will define as early offer that means if i am sending an invite message to you and i'm also sending the sdp at the same time that is called early offer i made the offer to you very early stage right whereas if i send the invite message then i wait i wait for like 150 seconds or whatever to see if you are gonna send me something if you don't then i'm going to offer you my sdp or my negotiation parameters that is called delay offer because i delete it so remember that if i send that invite and sdb together i am giving you all the uh i'm telling you what the as deep is going to be i'm not waiting for you to send me anything but if i wait for you to send me anything that will be called delay offer delay offer is illustrated below in delay offer the session initiator does not send its capability as you can see invite is sent there is no sdp until somewhere right about in middle in step number eight and nine so that is delay because the calling party waited x amount of time it so delay offer will wait calling party will wait for the call device to send his capability first list of kodak or whether it's an audio call or a video call now delay offer is recommended for sip trunk because it enables itsp service provider to provide their capability first so that you as a customer they can negotiate properly because obviously sometimes we don't have the power over service provider to decide what kodak or what to offer so we've got to wait for them to negotiate with us call manager will allow administrator to select which offer you want away may use cisco gateway supports both method but originating gateway by default is set to early offer call manager sip trunk by default set to delay alpha now early offer which is illustrated as you can see that is sit invite an sdp happens on step number one and two so right away so therefore all the capability is negotiated by send originating party okay in this case i get to dictate what you want to negotiate and early offer is enabled by default on a voice gateway i'm talking about cisco router acting as originating gateway but when you're establishing a cisco sip trunk from call managers ucm to any other device that safe trunk by default will have early offer as a delay upward enable so let me go show you how to configure a sip trunk that enables dna offer or earlier for you so we're going to go to our lab we're going to log into our call manager so i'm going to create a sip trunk and first i'm going to show you how to enable jelly offer so i'm going to go device i'm going to add a trunk this is a basically a just a demo trunk not going to work i'm going to say sip protocol is going to sip next and i'm going to call this demo sip trunk put a device pool now if see this option right here media termination point required if that is unchecked then most likely you have delay upper now when i say most likely is because this also depends on your profile so below at the very bottom you'll see there's something called sip robot right there that zip profile can also have delay offer and early offer enabled so that's why i say maybe but if i check this it's going to guarantee it's going to use earlier for every time a call passes through this trunk that is a must okay other even if the call needs an early offer or not okay so that some so every call that even does not need early offer capability will use uh early offer so what happened if you want to say no i want a solution that enable rnd opera only if the call needs it otherwise use delay offer so i'm going to keep that unchecked for that but to enable that feature first thing i'm going to do is i'm going to go back and create a c profile so i'm going to go to c profile and i'm going to copy this sip profile called vbs standard save profile vbc and when i'm there you'll see that there is an option here on this profile right there rd offers support for voice and video call it's disabled by default i want to say best effort do not insert mtp mandatory insert mtp which is media termination point only if needed so i'm going to basically specify a profile that says use m media termination point for our upper rd offer only if you didn't need it so i'm going to save that and i'm going to go to go back to the trunk let's call this sick trunk again i didn't save it i'm going to call this demo say trunk toronto and all you have to do at the very bottom you will see called save profile i'm going to select the bbc and save and that's how you enable early offer and delay upload so again this is not something that hasn't been will going to work because that's just a demo environment so as you can see this is basically the difference between them both early offer and delay offer hopefully you got an idea about what ship is how signaling works the type of different signaling the differences between early upward guillot analyzing the ship uh packets sdp you need to understand what those you know letters mean so you can easily uh troubleshoot and hopefully got understanding the process of paul being initiated and what stage it goes through so thank you for watching this video and i will see you in the