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CBT Nuggets

Troubleshoot Call Setup & Media Establishment

This Skill, led by Lalo Nunez, focuses on troubleshooting call setup and media establishment within Cisco Unified Border Element (CUBE) environments. It covers various SIP call flows, including normal call setup and teardown, and scenarios involving SIP responses like 404 Not Found, 487 Request Canceled, and 488 Not Acceptable Media. The Skill also delves into using CLI commands for debugging and verifying media parameters, and addresses common issues with inbound and outbound calls to ITSPs.

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58m 11 Videos 15 Questions

Skill 6 of 6 in Cisco H.323 and SIP

Introduction

Join Lalo Nunez as he completes Domain 1 of the CLACCM (Implementing Cisco Advanced Call Control and Mobility Services) blueprint. In this introduction video, Lalo discusses the topics we will cover in this Skill.


Normal Call Setup and Tear Down

First, we review a sip call flow with our normal call setup and teardown. Knowing what a successful or normal call setup and teardown should look like will allow us to easily identify when there is an issue with a sip call flow.

Below are the file(s) referenced in the video.

Knowledge Check

An Early Offer is when the media parameters are negotiated within the 200 OK and ACK within an SIP call flow.

  1. A
  2. B

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404 Not Found

Next, we examine a SIP call flow where we receive a 404 Not Found SIP response.

Below are the file(s) referenced in the video.

Knowledge Check

In this SIP Call Flow why did CUCM send back a response 404 Not Found?

  1. AThe called number was not configured within CUCM
  2. BThe CUCM server was offline
  3. CThe CODEC negotiated was not found
  4. DThe called party never answered the phone

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487 Request Canceled

Next, we examine a SIP call flow where we receive a 487 Request Canceled SIP response.

Below are the file(s) referenced in the video. Please note the .txt file(s) contain the output of the debug ccsip messages command.

Knowledge Check

What would cause a CANCEL SIP request to be sent while a call is attempting to be established?

  1. AThe call was terminated before a session was established
  2. BThe called party did not answer
  3. CThe called party was routed to voicemail
  4. DThe call was terminated after a session was established

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Scenario 1: Incoming calls from the ITSP are not successful

Next, we examine a SIP call flow where we cannot receive inbound calls from the environment.

Below are the file(s) referenced in the video. Please note the .txt file(s) contain the output of the debug ccsip messages command.

Knowledge Check

What header, within the SIP 404 Not Found, did we see the message "No matching outgoing dial-peer"?

  1. AWarning
  2. BInfo
  3. CAlert
  4. DException

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Scenario 2: Incoming calls from the ITSP are not successful

Next, we examine another SIP call flow where we cannot receive inbound calls from the environment.

Below are the file(s) referenced in the video. Please note the .txt file(s) contain the output of the debug ccsip messages command.

Knowledge Check

In this demo, what SIP response did we receive when our SIP Request eventually timed out?

  1. A408 Request Timeout
  2. B404 Request Timeout
  3. C408 Request Invalid
  4. D404 Request Invalid

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Scenario 3: Outbound calls to ITSP are not successful

In our next scenario, we cannot make outbound calls to the ITSP.

Below are the file(s) referenced in the video.

Knowledge Check

What was the "SIP Trunk Status" that CUCM displayed when our SIP Trunk was misconfigured with the wrong port?

  1. ANo Service
  2. BConnection Lost
  3. CLost Link
  4. DLink Status Failed

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Troubleshoot Media Establishment (CLI commands)

In this Nugget, we look at some CLI commands that help us look at the SDP offer/answer between call legs when a call is established.

Please note that I did not review the "debug ccsip messages" command, as we have covered the output of this command in past Skills. Nevertheless, "debug ccsip messages" should be one of the commands you should also be using in your arsenal.

Knowledge Check

Please reference the image above. The output shown is from which command?

  1. Ashow call active voice compact
  2. Bdebug voip ccapi inout
  3. Cshow call active voice brief
  4. Dshow call active voice summary

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Troubleshoot Media Establishment (The approach)

In this Nugget, we look at some factors we can examine when troubleshooting media issues with CUBE.

Knowledge Check

True or False: With Media Flow-Around signaling packets do not terminate on CUBE.

  1. A
  2. B
  3. C
  4. D

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Scenario 4: 488 Not Acceptable Media

Next, we look at a SIP call flow where we receive "488 Not Acceptable Media".

Below are the file(s) referenced in the video. Please note the .txt file(s) contain the output of the debug ccsip messages command.

Knowledge Check

In this demo, what command was used to show us the dial-peers used to set up the call and the negotiated codecs?

  1. Ashow call active voice brief
  2. Bshow call active voice all
  3. Cshow call active voice summary
  4. Dshow call voice

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Validation

Welcome to the Validation section. We will be “validating” your knowledge through a series of questions. This assessment will gauge your understanding of the content within this Skill. Some of these questions could be associated with visual aids such as images or videos. Watch the Validation video, where we review the answers.

Good luck, and you got this!

Note: Please complete your quiz questions before moving on to the next Skill.

Please refer to the images above to answer the questions below. The output is taken from CUBE for the same active call.

Knowledge Check

Which is the calling number?

  1. A2401
  2. B17735551234
  3. C1890
  4. D1880

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Knowledge Check

What is the called number?

  1. A17735551234
  2. B2401
  3. C1890
  4. D1880

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Knowledge Check

What port is being used for media by Pedro's endpoint (which has an IP address of 10.0.3.104)?

  1. A29358
  2. B16402
  3. C8022
  4. D8020

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Knowledge Check

What are the inbound and outbound dial-peers being used?

  1. Ainbound: 100 outbound: 201
  2. Binbound: 201 outbound: 100
  3. Cinbound: 605 outbound: 606
  4. Dinbound: 606 outbound: 605

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Knowledge Check

Where do we see which CODEC was negotiated?

  1. Ag711ulaw
  2. Bg711alaw
  3. Cg729r8
  4. Dilbc

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Knowledge Check

What is the local port CUBE is using for media for the call between CUBE and Pedro?

  1. A8020
  2. B8022
  3. C29358
  4. D16402

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View Transcript

Introduction

0:00Welcome to this skill, "Trommishik Qa'al setup and media establishment."

0:03So in this skill, we look at several different Qa'al flows where we are

0:06receiving different

0:07responses when we're trying to setup a Qa'al. We also have a discussion on how

0:11we can approach

0:12troubleshooting media establishment. So let's see what else you want to recover

0:16in this skill.

0:16So in our first video, we'll have a review of a normal Qa'al setup and teardown

0:22.

0:22Then finally, now we'll look at a sip Qa'al flow, we'll receive it for the

0:26found,

0:28along with a 487 request canceled. Then we have three different scenarios where

0:32either inbound or

0:33on-ball calls to the ITSP are not functioning. After that, we'll see that we

0:38can execute

0:39certain commands on a command line and then we look at some factors that may

0:43come into play

0:43when it comes to troubleshooting media establishment. Then finally, we'll look

0:48at our

0:48final scenario where we'll receive a 488 non-assetteable media. So as always, I

0:54thank you for allowing

0:54me to be part of your journey and I'll see you in this video.

Normal Call Setup and Tear Down

0:00So in this thing, it will go over a normal cost set up and turn it up.

0:03Now you've been with me throughout this course, this is a purely review.

0:06So we have a scenario where we have an inbound call from the ITSP to Pedro.

0:11So in a normal cost set up and turn it on, we have our request here,

0:15that being the invite. Then we have a 100 shrine, 180 ring, and 200 roll cap.

0:21These are our responses. Then an ACK, which is a request, is sent indicating

0:28confirmation that it received a final response to the invite request.

0:34Then we have media that is later established and eventually someone

0:36terminates this phone call. So here's a buy request.

0:40Then later we get a 200k, which is our response to that request.

0:45So this is the foundation here of a basic slip-call flow.

0:49And this is important, especially when looking at slip-call flows, attempting

0:52to

0:52determine what the issue is. This is exactly what we see in our

0:56call flow here. So this is Q, this is CUCM, this is Pedro, and his endpoint.

1:04So here we see invite, we see 100 shrine, 180 ring in, and 200 roll cap.

1:10Then finally we see our acknowledgement, which is confirmation that we received

1:14the final response, this being the 200k, to the invite request.

1:19With this invite, we see SDP information that we now know is

1:24the indication of early offer. So the media parameters, like the codecs,

1:28that we're going to use will be negotiated in the invite,

1:32and later we get a response in a 200k.

1:36So here we also have an invite being set to Pedro's phone,

1:39which is CUCM. And we can see CUCM sends an invite here,

1:44but there's nothing within the body. So this gives us an indication that this

1:48is

1:48the late offer. So with the late offer, the media

1:52parameters, like the codec, is negotiated in the 200k,

1:57along with the acknowledgement. So that's for you would find your SDP

2:01information. But let's do a live warsar capture here.

2:07So what we do is make a phone call from Pedro's phone.

2:10So let me start the capture here, but we're going to filter only SIP messages.

2:16So we're going to go off hook, we still notified there,

2:20the VDAL, out to the HSP. We see an invite 100 try and 180 ring.

2:26Now when I enter a call, so we initially saw the invite,

2:32the 100 try and the 180 ring in. Then once we entered the phone call,

2:36we saw the 200k, which was the final response to our invite.

2:40And then we saw our acknowledgement. And there we go, we have a basic review of

2:45the

2:45normal call setup and teardown for a SIP call.

404 Not Found

0:00So, that's the other scenario, we have an inbound call from the ITSP, and I

0:05call it

0:05"Rotted to CCM", and eventually we get this "44" not found.

0:09So, let's bring over that call flow.

0:12So, we get an invite here.

0:15The number that the ITSP is trying to dial is 1,813-555-6523.

0:18So, we get a try back from CCM, so this tells us that we can't communicate with

0:26that CCM

0:27server.

0:28And then, eventually we get a "44" not found.

0:31So, let's go over to CCM.

0:35So, there's some over the call Pedro, there was a dial 1,813-555-2401.

0:41Now, in my environment, "241" is the "stentional Pedro".

0:46So, if we go to the device and phone, go to "fine", and we go to "Pedro's Phone

0:56", we

0:56see "2401" here, and you may ask, "Well, how are we transforming a translating,

1:021,813-555-2401,

1:062,2401?"

1:08So, you can do that on the cube itself using voice translation rules.

1:17And we looked at voice translation rules during serial core, and we'll revisit

1:23them later

1:24in this course.

1:25Now, I can also do the same using CCM by using lysing translation patterns.

1:32So if we go to "Car routing", go down to "translation pattern", I'll hit "fine

1:37".

1:37We have a couple here, but if I look at this one, for example, this will

1:42translate this

1:42number, the "x" being any numbers from 0 to 9, down to the "last" for "gets".

1:49So, this is how someone can dial 1, 813-555-2401, that being Pedro.

1:57See, some received that number, and with the translation pattern, it translates

2:01that to

2:01"2401".

2:02Now, I can see you see, we go to "Car routing", and go to "Rough planning port

2:09", and I can

2:09say "show me any patterns that match 2, 401".

2:12I would find, then we see a "dural 3" number here, which belongs to Pedro.

2:18As we can see, right there.

2:22So how about this number, 1, 813-555-6523?

2:27So based on the information that we just went over, I would expect to see a

2:31directory number

2:32somewhere in my system for 6, 5, 2, 3.

2:35So going back to CCM, let's go back to "Car routing", "Rough plan report".

2:42Let's do 6, 5, 2, 3.

2:43I would find here, and we can see that I do not have that directory number

2:48configured

2:49within CCM.

2:52And now we know the reason, but we received this 404, and I found from CCM,

2:57because CCM

2:58is telling you, I do not have this number within my CCM database.

3:04So in fact, let's bring up "washer", I have my phone here.

3:09Going to dial 1, 1, 2, 3, 5.

3:12I had a call here.

3:13"Your call cannot be completed, it's dialed.

3:15Please consult your directory and call again."

3:18So notice what happens if I wait here.

3:21Let me end up all, and I'll stop that wireshark capture.

3:29So we can see here that in by the sentover to CCM, we get a 100 shrine, we get

3:35the 183

3:36session progress where you play some media about a call being invalid.

3:40But I did get this 404 now found because I waited for the announcement to

3:43finish.

3:44What's what happens if I repeat this?

3:46What do not wait for the announcement to play out?

3:49So let me go ahead and start a new capture.

3:53I'll here read that.

3:54"Your call cannot be completed, it's dialed."

3:56And this time I'll end call.

4:00So let me stop the capture there.

4:02So notice here, we get a 487 request cancel, which we'll talk about in later n

4:07ugget.

4:08We did not get a 404 now found.

4:12For the reason being, we terminate the call during the 183 session progress

4:17when a media

4:18was being played to us before the fact that we would see a 404 now found.

4:24So in this nugget we went over a scenario where we received a 404 now found.

4:28And we saw the reason that we did get the response because CCM did not have

4:32that number

4:33within it. See you in the next video.

487 Request Canceled

0:00So next we'll take a look at a sip car flow that has a 47 request cancel. In

0:04this scenario,

0:05Pedro makes it a phone call and later terminates that phone call.

0:09So Pedro is attempting to make it a phone call.

0:13So the question here is the call established or not. So did the destination

0:21answer the phone call.

0:22So Pedro did establish that phone call.

0:26So the other side did enter the call. So if the answer is yes here,

0:30then at some point Pedro determined that phone call we would expect to see a

0:36bite.

0:36And that's not what we see here. So the call wasn't established.

0:40So the destination is the example still ringing. And Pedro terminates that

0:45phone call.

0:45So at this point a cancel will be sent.

0:48So it really comes down to what the call established.

0:52In both cases Pedro did to remain the phone call. But if the call was indeed

0:57established we get a bite.

0:58If it wasn't established yet at that point we get a cancel.

1:01So taking a look at that call flow we have an invite. We have one who trying.

1:07We have 180 bringing here.

1:09So you can see the support for Prag here. But eventually we get a cancel.

1:15Now this call flow is from a perspective of Q.

1:19So we don't see the messages from the phone over to CUCM.

1:23But as soon Pedro hit and call at that point a cancel was sent from the

1:29endpoint over to CUCM.

1:31That's exactly what we see here that sent over to Q.

1:34Then we get a 200 okay to that cancel. Then we get a final response to the

1:40original invite that was sent right here.

1:42And the final response is 487 request cancel.

1:47So let's go back and do a live capture once again. I will hit start here.

1:52Let's go ahead and bring up our phone. I'm going to give the ITSP a call here.

1:58Now watch what happens when I hit and call what that phone is to ring in.

2:07So here we can see that the cancel was sent from this IP phone. It has the IP

2:14address 1012 CUCM.

2:16Then we see our final response that being a 47 request cancel.

2:21Let's run that test once again. I'll start new capture.

2:25I'm going to hit redot here.

2:28But this time I'll insert,

2:30put the call mutual voice feedback. Now watch what happens when I hit and call.

2:35Then we stop the capture now.

2:38So now we see that a buy was sent from the endpoint over to CUCM.

2:44For the reason being the call was already established.

2:46So in fact we have another call flow here.

2:51So this call flow is very similar. We have an invite on the shrine when it is

2:55ringing. We have a 200k.

2:57Followed by the acknowledgement. So this tells us that the session was

3:00established.

3:01So at some point Pedro terminated the phone call and then sent a buy over to C

3:05UCM.

3:06So in this again we looked at a sit call flow that contained a 487 request

3:11cancel.

3:12And the reason for this response because one side, in this case Pedro,

3:17terminated the call before it was established.

3:19[BLANK_AUDIO]

Scenario 1: Incoming calls from the ITSP are not successful

0:00In our next example here, incoming calls from the ITSP are not successful.

0:04So when we have an incoming call into my environment, the OutPair100 is the in

0:10bound

0:10DOP here, and then the OutPair 102 is the Aband DOP here.

0:16So here we see the DOP here of 100, followed by the DOP here of 200.

0:20Here we have the IP address of Q, and also shown here is the IP address that's

0:26defined

0:27in our script from within CCM.

0:30I have a YouTube here, I do have login enabled, I'll go ahead and include my

0:34log, we'll do

0:35a debug CCM set with messages.

0:39So what we're going to do is make an inbound call into our environment.

0:42So I would attempt to call Pedro, I head down there, and we'll see what happens

0:47here

0:47in a second.

0:48So you might not hear that, but that's the tone that we get.

0:54Let's go ahead and send that log over to our TFTP server.

1:00We'll name this inbound1.txt.

1:08So here's a conversation between ITSP and Q.

1:11And you see here this 404 not found.

1:14Now previously we saw this when the directory number is this within the CCM

1:18database.

1:19What we see here at the DOP number is 1A13-555-2401.

1:26Let's go to CCM.

1:28We can verify that directory number doesn't exist.

1:30I should show you I did not delete the offline before I had record.

1:35It's a call routing, route panam report.

1:392401.

1:40And here we see the endpoint for Pedro, therefore line 1 has 2401.

1:47So you know that in this case the directory number does exist.

1:52So we see the ITSP, we see Q, but notice we don't have any communication here

1:57with CCM.

1:59So what happened here?

2:01So let's take a look at this 404 in our found.

2:04And this will have some additional piece of information within the warning

2:08header here.

2:09It says no matching outgoing DOP here.

2:12So remember I said this call lag, the ITSP over to Q, we use DOP here 100.

2:20And then the DOP here from Q over to CCM should use the DOP here, 102.

2:28So look at this, we see DOP here, 100.

2:30This is the inbound DOP here.

2:35Then DOP here, 102 is the ABAN DOP here pointing to CCM.

2:39In fact we see the IP address of our subscriber, or I should say our series in

2:43publisher there.

2:46Now maybe your ISPite upon the fact that this says shutdown.

2:51When a DOP here is shutdown, it cannot be used by Q.

2:54So let's go ahead and validate that is the fact here.

2:57Let's go to Q, let's do a show run section, DOP here voice, I need my space

3:04there.

3:05Let's look at 102.

3:08So indeed we see the shutdown keyword there.

3:11And this DOP here will be used from Q over to CCM for the ABAN DOP here.

3:20But we see from our CIP call flow, we get a 404 and within that 404, we do have

3:26a warning

3:27header that states no matching outgoing DOP here.

3:31And that would be correct for the reason being DOP here, 102 is shutdown.

3:36So let's go back over to Q.

3:39Let me clear my screen there.

3:40We do a config T. We'll do a DOP here.

3:45Voice 102.

3:47We'll do a no shutdown.

3:49So now we look at that DOP here once again.

3:53So at this point this DOP here should be active.

3:55So now I'm making a test phone call once again from the ITSP over the schedule.

3:59This phone should ring.

4:01And it does.

4:03Let's put a mute there.

4:05So let's go back to Q.

4:09So let's see what DOP here is being used by doing a show call active voice

4:12brief.

4:12And we're going to include the peer ID.

4:15So we see here we have DOP here, 100 as the AMBON DOP here and DOP here, 102 as

4:20the ABAN

4:20DOP here.

4:22Since the DOP here was shut, that's the reason that we received a 404 and I

4:28found in our

4:28CIP call flow with the warning header once again of no matching outgoing DOP

4:34here.

4:35So once again we went over DOP here during the CIP call.

4:39So we should have some background or be familiar with DOP here at the moment

4:44and we'll further

4:45discuss DOP here later in this course.

4:48What the takeaway is is that we saw that 404 with the no matching outgoing DOP

4:53here and

4:54it should make you think which DOP here is being used for that call lag for the

4:59ABAN

5:00call lag between Q and CUC up.

5:03And the finally nugget will look at another scenario where incoming calls for

5:07an ITSP

5:07are not successful.

5:08>> Okay.

Scenario 2: Incoming calls from the ITSP are not successful

0:00So in our net scenario, we have the same issue incoming calls for the ITSP are

0:03not successful

0:05and you'd like the previous niggas, the inbound up here for cube is down here

0:09100

0:10and the upbound up here from cube over to cu7 is down here one or two.

0:15So here we have the IP address of cube and also the silt trunk that's

0:20configured within cu7

0:22also has the IP address 10.0.3.2.2.3.

0:27Alright, so my debug is still turned down from the previous nugget. We'll go

0:30ahead and clear the log.

0:31Let's go ahead and make that call inbound to our environment one more time

0:36and after a second, we can hear that tone in the background. So let me call.

0:43Alright, so let's send that out back to our TFTP server when in this inbound

0:50two, that THC.

0:54So as we already know, here's the ITSP, we have cube and here we have cu7.

0:59So we have an invite from the ITSP that sent the cube,

1:04cu7 is back over on a train as we see here and then we see an invite sent to cu

1:107.

1:11Now that this communication is one way, so we don't receive any communication

1:16back from the server.

1:17Eventually what happens here is that we get a 4.08 request timeout.

1:23So the reason here we got the 4.08 request timeout is because cube was unable

1:29to get every response back from the server.

1:31So looking at our configuration here, we have our inbound.op here

1:36and we also have our abound.op here.

1:40The server that's now responding has the IP address of 10.0.4.55.

1:46In this diagram here, what is the IP address of our cu7 server? It is 10.0.4.0.

1:5250.

1:54So this .op here is doing a job. It's being used as the album call link and

1:57running the call to this IP address right there.

2:00But this IP address, at least in my environment, does not exist. So in order to

2:04correct the issue,

2:05we need to go back to the .op here. So let's go to config team.

2:10Let's go to .op here, voice 102 and I fact finger that and there we go.

2:20And in fact, before I make the change, let's go ahead and verify the current

2:22config of that .op here.

2:24So we can see here we have the wrong IP address.

2:27So now let's go back to .op here. We'll go ahead and do session target

2:33ipv4 and then we do 10.0.4.50. We'll hit enter.

2:40So now we should have the correct config here, which we do.

2:46So now what happens if I hit readout once again, we'll pet just for a ring.

2:50And we can see that is the case.

2:52So in this again, we saw another scenario where within the call flow, we got a

2:59408 request

3:00timeout for the reason being that the album call link from Q out to CECM was

3:06referencing the

3:06wrong IP address. Once we made a change to the appropriate .op here, at that

3:10point, we're able

3:11to write that call correctly.

3:12[BLANK_AUDIO]

Scenario 3: Outbound calls to ITSP are not successful

0:00So this time we have a scenario where we're dealing with album calls.

0:03album calls to the ITSP are not successful.

0:06So for an album call, the inbound dial peer is going to be dialed up here 100,

0:12and the album dialed peer to the ITSP will be dialed here 201.

0:17So once again we have our IP address, our Qube, and also our CUCM configuration

0:22, etc.

0:25Let's go ahead and see what happens when Pedro makes a phone call out to the IT

0:29SP.

0:29So I'm going to choose a number here, that number being 1773-55-1234.

0:36Let's see what occurs here.

0:39And that's the tone that we hear.

0:43So let's go ahead and do a live packet capture here.

0:47So this packet capture is taken from the perspective of Pedro's phone.

0:50So let me start it.

0:52I'll continue without saving.

0:56Let's go ahead and make that phone call one more time.

0:58So we see a 100 train, it hangs there, and eventually we get a message here,

1:07and the response or the status is 480, temporarily not available.

1:12So in fact, if we look on Qube, or I did have a debug turned on, if I do a show

1:16log,

1:17we don't see any entries, so this tells us that we're not even hitting our Qube

1:24.

1:24So for kicks, let's do a packet capture on CUCM.

1:27So we'll start a capture here.

1:29We'll name this outbound test one.

1:32We'll hit enter there.

1:34We're for that to start.

1:36So we'll make that phone call once again, and the same should occur that call

1:40should

1:40not complete, and we should hit our tone soon.

1:43Here we go.

1:45So I'll hit control C to stop that capture.

1:48Let's go ahead and transfer it to our secure FTP server, the name being out

1:53bound.

1:54That's one.cap.

1:57So at this point, we'll enter in the information for our server, and we'll be

2:01for our transfer

2:01to complete.

2:02All right, so our transfer is completed.

2:07And what we see here is the communication between Pedro's phone and CUCM.

2:13So we still see this 480 temporarily not available.

2:17What we would expect to see here is some communication with 10.0.3.253, which

2:24is the IP address of our

2:28Qube.

2:29So CUCM communicates to Qube over a SipTron.

2:32So let me go to device and trunk.

2:36So here's our SipTron over the Qube, and I do have options ping turned on,

2:41which CUCM

2:42uses as a keep a light mechanism, like a heartbeat.

2:45Are you there?

2:46Are you there?

2:47And for SipTron status, we see no service, and it's been out of service for

2:51about nine

2:52minutes.

2:53So let's go to SipTron and if you scroll to the bottom there, there's a portion

3:00of the

3:00disk configuration where we configure the IP address.

3:04So we see here the IP address, 10.0.3.253.

3:09And going to Qube, I do a show IP interface brief, and now I just go to the

3:14sign that

3:15is the same IP address that we see here.

3:17If I were to ping that CUCM server, we can see that we can communicate with CUC

3:24M successfully.

3:26But if you're looking carefully, we see a destination port of 5090.

3:31Say by default we'll use 5060 or 5061, depending if you're doing a secure step.

3:37So obviously this port isn't correct.

3:42So going back to my slide here, that port's really hard to see, but it does

3:47save 5090.

3:49So let's go ahead and correct this.

3:50We'll put the port 5060, we'll hit save, and then we'll go ahead and do a reset

3:55on this

3:56trunk.

3:57So we'll go back to the bison trunk, and we'll give this a moment here to

4:01refresh.

4:02So I'll wait about one minute or so, and then refresh the screen.

4:08Alright now we see for our CUCM we are in full service, and time in full

4:12service, zero

4:13minutes because it just came up.

4:16So now if I were to make a call once again from Pedro over to the ITSP, I'd

4:20read that.

4:21Let me see the call is successful.

4:27So going back to Qube, let me clear my screen there.

4:30If I check what dial pairs are being used, the dial pair between CUCM and Qube

4:36is the

4:36inbound dial pair of 100, and the outbound dial pair out to the ITSP is dial

4:43pair 201.

4:44So in this nugget, you and I look at the scenario where outbound phone calls

4:47were not

4:48successful.

4:49So even though we had the correct dial pairs in place, we were receiving a 480

4:54temporary

4:55now available.

4:57And now it's because within CUCM for our CIPChunk configuration, we had the

5:01correct IP address

5:03with the incorrect port to find.

5:04[BLANK_AUDIO]

Troubleshoot Media Establishment (CLI commands)

0:00and this nugget will go over troubleshooting media establishment and we'll do

0:03so from the

0:04perspective of Qube. So we can use these commands for many reasons but in our

0:10case for this nugget

0:11we want to ensure the media information that was negotiated. So let's go over

0:17to Qube. So what you

0:18and I will do first is pin up a call. So let me have a call from the ITSP over

0:23the pejorole.

0:24So this phone should be here shortly. We'll answer it. We'll go ahead and put

0:28both calls on you.

0:30And the first command we'll show is show call active voice compact.

0:36So first thing we see is the quality. So this should be the inbound

0:43quality and then the outbound quality for this call. We have answered here and

0:48this would be our

0:49quality number. We have originated here and this would be our call number. But

0:55we can see here

0:56the code that was negotiated by being G7 11. This T 41 tells us the duration of

1:03this call in seconds.

1:04So keep in mind this is an active phone call. So if I were to repeat that same

1:08command,

1:09we see that number increased. We see peer address which is the phone number

1:14involved in each party.

1:16Then we have the IP address and the port number for the RTP stream. In fact, if

1:21we go to CUCM,

1:22we see that prejudice phone has the IP address 104, which is the same with

1:28p address that we see here. It would be browse to his phone and go to stream

1:34one.

1:34We see for local address, we have the IP address of prejudice phone and that

1:39port of

1:4022,330. That's the port that we see right there. Now for remote address, we see

1:47the IP address of Q

1:49but port 8,014. And we'll see 8,014 when we look at a different command. So let

1:56's do a show,

1:57voice of RP, RTP connections.

2:04Let me see source in this nation called IDs. But here we see the local RTP port

2:10from the perspective

2:12of Q. We saw in the previous command that the app on call egg was called ID 227

2:18.

2:19So going back to our show voice of RP RTP connections, we see that call ID 227

2:27is using

2:28the local port on Q for 8,014. And that's the same port number that we see

2:33right there.

2:34In fact, let me go ahead and decrease the font just a little bit.

2:39So I was trying to decrease my font so we can align these columns. But here we

2:45can see the

2:46local port along with that remote port. So we see 22,330. That's also the port

2:53that we see right there.

2:55So going back to Q, let me go ahead and clear my screen there. Let's do a show

3:01call active voice

3:04brief. Let me scroll down a bit. And I want to focus on this body of touch

3:10right here.

3:11So once again, we have call ID 226 227. So we have the PR ID. So this is dialed

3:18here 100

3:19for the inbound dial here. Then we have dialed here 102 for the onbound dial

3:25here.

3:25So here we have the calling number. Then here we have the call number. So we

3:32have the IP address

3:33of the IP S P of the endpoint that being federal. But here we can also see

3:37which codec was being

3:39negotiated. And this case being G7 11 U law. These are three commands that we

3:46can use to verify

3:48the media that was negotiated. And we also have debug voice to IP CC API in out

3:54. This command

3:55here traces the entire path of the call from beginning 10. And it does so via

4:02the call control API.

4:04So offline, I went ahead and ran that debug and I exported to a file via TFTP.

4:13But you can get a

4:14good indication of how much information is in this file. Now keep in mind this

4:19was an inbound call

4:20from the IP SP over the Pedro. And you can see a lot of information here

4:25breaking down the

4:27execution of the whole entire call. So at this point, my goal is not to show

4:30you how to read

4:31this from beginning 10. Let me focus on a certain portion of this debug. So

4:37here, Q receives the

4:39capabilities from this call leg of 192. And we can see a codec specified here.

4:46Later,

4:46Q acknowledges this. This is coming from call like 191. Well, we can see the

4:52codec that we are

4:53acknowledging that being G7 11 U law. So once again, at this point, don't be

4:58too overwhelmed

5:00from the output here. My goal was to show you with these commands, we're able

5:05to see what media

5:06was negotiated for a call that was established. And then finally, nugget, we'll

5:11go ahead and examine

5:12other factors that we can look into when it comes to troubleshooting media

5:17establishment. Thank you.

5:18[BLANK_AUDIO]

Troubleshoot Media Establishment (The approach)

0:00So next, we'll look at other factors that may come into play when it comes to

0:04troubleshooting

0:04media within Qube.

0:06So these are questions that we should be asking.

0:10And some of these questions we answer with the appropriate debug or show

0:15command.

0:16First question being, does this issue happen randomly?

0:18If the answer is no, then we easily could reproduce the issue, which makes it

0:23easier

0:24to get a packet captured, for example.

0:26We could also use the appropriate debug and show commands to help us troublesh

0:31oot the

0:31issue.

0:32If the answer is yes, it does happen randomly, that makes your job a little bit

0:37more difficult

0:38because oftentimes we need the event to happen in real time so we can collect

0:42logs.

0:43So you might have to enable the spanning of a port, spainby and acronym for

0:47switch port

0:48analyzer, and then hope that we can capture the event happening.

0:52Next, we should grab a detailed description of the issue.

0:56Is the problem making an inbox phone call or a bound?

1:00Is it calling a specific number, for example?

1:04Is the issue about the quality of the voice?

1:06Is it choppy?

1:07Is there a message that was played at some point during the call setup?

1:12Knowing somebody's answer is we hope you narrow down your troubleshooting or at

1:16least

1:16give you an idea where you need to big deal the troubleshoot.

1:20Our next question is the media, the RTP, flowing through Q.

1:26We have media flow through.

1:29We also have media flow around.

1:35With media flow through, which is the default in many cases, the media and the

1:40signaling

1:41packets terminate on Q.

1:44The media flow around as the name indicates, the signaling terminates on Q, but

1:51the media

1:52bypasses Q entirely and flows between source and destination.

1:56So in this case, if the answer to this question is no, then Q has nothing to do

2:00with the media.

2:02So any of you that we do have a problem with the media, we have to focus on

2:06source and

2:06destination, those endpoints and the networks in between those.

2:11So let's put devices involved in the call flow.

2:15This is good to get the entire picture or understanding of the call flow from

2:19beginning

2:19to end.

2:20So we can use debug and show commands once again to verify the IP addresses

2:26along with

2:26ports that are being used in the media path.

2:31Sometimes you may find it the wrong IP addresses and ports are being referenced

2:35.

2:35So next, which codec is being negotiated?

2:39So let's say you have remote sites, there are a number of nowhere.

2:44Any sites have low bandwidth and for that reason, we're using G729.

2:48And once again, with the appropriate show commands, if we find out that we're

2:51actually

2:52doing G711, especially over those low bandwidth links, at that point, that may

2:58explain why

2:59we have poor media quality.

3:01So now, so there are any ACLs or active controllers being used.

3:05And if so, are we blocking certain IP addresses or ports?

3:10And therefore, that's the reason that the media is not being established.

3:13Are there any firewalls involved in this path?

3:18Just like access control lists, if we're blocking certain IP addresses or ports

3:22, that

3:23definitely can cause audio issues.

3:26The Vining are the media bindings correct.

3:29And it should say, "Duh, not there."

3:31So when a media binding, we're binding an interface on cube and that interface

3:37has an

3:38IP address associated with it.

3:39And this is the IP address that cube advertisers.

3:42So for example, if the interface is 10.10.10.100, well, cube also has another

3:49interface, 10.11.101.

3:53This IP address of 101 should be the IP address advertised.

3:58For the reason being, our devices can route to this IP address.

4:02The binding was to this IP address and the devices cannot route to that IP

4:07address.

4:07And then at that point, we would have issues with the call set up and also with

4:12the media.

4:13And I'm going to be the first to tell you that the troubleshoot media is

4:16definitely a

4:17challenge.

4:19And once you see here, some of the questions that you may ask to narrow down

4:22your troubleshooting.

4:24I'm confident I won't fulfill an entire course just with troubleshooting media.

4:28But here, asking the right questions and knowing what commands that we should

4:31execute to find

4:32out further information will definitely give you a good starting point.

Scenario 4: 488 Not Acceptable Media

0:00So here in our last scenario, all hebound calls for the ITSP cannot be

0:04completed.

0:04In fact, here we see in the sip call flow that was provided, we've received a 4

0:0988 non-acceptable medium.

0:11All right, so let's verify this. We'll go ahead and do a debug CC sip messages.

0:17I do have login turned on, so let me clean my log. Let's bring up our phones.

0:23I'm going to hit readout here in the ITSP and see what happens.

0:29I hope you can hear that tone in the background.

0:31So that I've called it not complete. So let's go ahead and send that to our TFT

0:36P server.

0:37So you see invite me Q and everything seems to be pretty normal. We'll be look.

0:44The ITSP is offering D711. So from Q, we get back a 488 non-acceptable.

0:53So from Q, we get a 488 non-acceptable media. So we saw that the ITSP was

0:59offering G711.

1:00But Q for whatever reason did not like that and send back the 488.

1:05So in my environment, the dial up here, or I should say the inbound dial up

1:10here,

1:10that is used is dial up here 100. So let's go ahead and take a look at the dial

1:15up here.

1:16We'll do show run section dial up here voice 100.

1:22And we see a configuration here. What the dial up here is configured for an

1:26advanced audio coding

1:28codec. So what's occurring here is that the ITSP is offering G711 in its

1:33negotiation.

1:34So I put ITSP here. But Q based on the dial up here configuration,

1:41is only configured to accept this advanced audio coding codec. So therefore, we

1:46don't have a match

1:47here. And that's the reason Q sends back this 488 non-acceptable media. So we

1:53can fix this. We're

1:55going to dial up here voice 100. And for codec, we're going to hard code the G7

2:0111 U-Law.

2:02For the reason being the ITSP told me that they only support G711. So with that

2:09in place,

2:09if I make that phone call once again, and read dial, what happens here, they're

2:14quite

2:14now successful. So if I do a show call active voice brief,

2:22let me just put just a little bit there. We can't see dial up here 100 for the

2:29inbound dial up here and dial up here 102 for the up on down here. And now we

2:33have negotiated

2:34G711 as I quote it. So in this, again, we saw a call flow where we received a 4

2:4088 non-acceptable

2:41media. And it's important to know what dial up here is that we are expecting to

2:45be used.

2:46So in this case, dial up here 100 for any cause coming from the ITSP. And then

2:50looking at the

2:51configuration for that dial up here, we saw previously that we were advertising

2:55a codec

2:56that was not supported by the ITSP. They're modifying the dial up here to

3:01advertise the

3:02correct codec, resolve their issues.

3:04[BLANK_AUDIO]

Validation

0:00Congratulations on making it to the end of the skill. Next you and I go to our

0:03validation where we answer series of questions

0:06We're forcing the knowledge that you will not obtain in this skill. Just get

0:09started and dive in

0:10All right, we can see here. We have several images that we can reference. So

0:15question one

0:15What is the calling number?

0:18So let's go to our first image here. This is so called active voice compact

0:22And we can see here that we have the

0:26Answer or the originated the answer is the calling number or calling party and

0:32The originated is the called party

0:35So based on this screenshot the calling number is two four zero one and the

0:39call party is one seven seven three five five five

0:42One two three four so here Pedro made a album call out to the I2SP. Let's look

0:48at the other screenshot here

0:51So we also see the answer here. That's for the two four zero one the calling

0:55party and the originate

0:57They mean the cloud party

0:59With the number that ends with one two three four

1:01So let's go back to our question. So the calling number is Pedro at two four

1:07zero one

1:08I hit submit. We also saw that the cloud number was the number out on the I2SP

1:14That's the number here that ends with one two three four and I hit submit there

1:19And that's question what ports being used for media by pejos endpoint

1:24And here we have a clue here that the endpoint has the IP address of 10.0 that

1:293.104. So let's go back to our images

1:31So here on this first one

1:34Here we have Pedro's IP address and a port of

1:372358

1:40Let's look at the other image here and we also see here for Pedro the port

1:45being used

1:462358

1:50And

1:52Here for show VoIP RTP connections

1:54We can see for pejos endpoint of 10.0.3. The 104 we also have port

2:012358

2:04So going back to our question our answer is 2358

2:09So next what are the inbound and not bond up here is being used

2:16So let's go back and we saw in a previous look it that would show call active

2:21voice brief

2:21We scroll down a bit this output would show us the peer ID

2:26We can see that the appear 100 is being used for Pedro who is a calling party

2:31That's the person that initiated the phone call

2:33Then now peer 201 is being used for the album calling out to the I2SP

2:39So this number here is the call party

2:43So our answer would be for the inbound up here. That's 100. I'm going to be up

2:47on the up here

2:48to one so let's go back to our question there and

2:51So the inbound up here is 100 and the album is to one

2:56And our next question where do we see which code it was negotiated?

3:02So let's go back up and let's look at our first screenshot

3:06We can see here that g7 11 was negotiated

3:11And we look at the show call out the voice brief

3:13We also see g7 11 there

3:17Then we show voice of IP RTP connections that's information that we don't see

3:23within this upload

3:24So going back to our question the answer is g7 11 you law right here

3:30So in our final question what is a local port cube using for media for the call

3:36between Q and Pedro

3:39So going back up to our images the output that we want to focus on is this show

3:44voice of IP RTP connections

3:46So here is the IP address of Pedro's endpoint

3:49And we can see here the port being used locally on Q is

3:538,020 so with that being said let's go down and choose that answer

3:588,020 and there we go so with that our data instruction is complete and I will

4:04see you in that skill

4:05[BLANK_AUDIO]

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