 Welcome to the Network Engineering Video Blog. I am your host, Michael Crane. In today's video, we are going to use Wireshark to look at a SIP 3-way conference call. The call flow will be A Call C, then B Calls A, followed by A Conferencing with B and C. We will be using our SIP test bed we built in video number 65 for this testing. So let's get started. All right, let's see here. So here's our call trace. Let's go ahead and look at the the call flow. So I'm going to go to telephony, wipe calls, and I'm going to highlight phone A and B. That's where we have our our mirroring ports mirroring. Okay, and we're going to look at the flow sequence. Okay, so just to show you here, so dot 44 right here is phone A, dot 46 is phone C, and dot 45 is phone B. Okay, so it's kind of it's kind of jumbled up in the and this call flow, but it should be okay. Okay, so phone A if you remember calls phone C, right, so here's the invite to phone C, and I'm not going to go through the standard calls and holds because I have videos describing how those work. So we're mainly interested in the conference call portion of it. And spoiler alert, there's not a lot interesting in SIP for conference calls and and we'll talk about it when we get to the end. Okay, there we go. Okay, so A call C, and I got the one under trying ringing and the 200 okay, phone A exit, and we've got two way voice pass setup, right? Okay, so then I had phone B right here, call A. And so it calls a right, and we get the one under trying and a ringing. Okay, so to answer phone B, phone A has to put phone C on hold, right? And so this is the hold and we can go verify that real quick here. Let me move this off the screen. And that's this invite. That's what's nice about this. If you click on any of these, you're going to watch it like that. If you click on any of these, you can see row light up or get highlighted. The selected message gets highlighted, right? While the row does. So here's the trying, ringing. Okay, we're interested in this. It's the invite. It's going to be a call. He's going to be putting them on hold. Here's the message body here. Come down here and you can see with this attribute right here, send only. So he's saying I'm putting you on hold. If you don't understand that, I have a video where I talk about call hold. Okay. All right here. So he puts, so phone A puts phone C right here on hold and phone C 200 acts back with a 200 okay and receive only. Yep, right here. Okay, we get an act. And now, now that phone C is on hold, now the user can answer the call from phone B. And that's what this 200 okay is right here. Okay. And of course, we get the act back from B and we have two way media. So now phone A wants to tie all the, all these calls together, right? So he's got a call in progress with phone B, right? He's got phone C on hold. And so when the user hits the conference button, the phone asks, okay, who do you want a conference with? When the user selects, oh, well, I want a conference with line one, which is phone C right here. The only thing the phone does is he takes them off hold. As far as SIP goes, right? So, and we can go look at this, this invite right here. And just move them off screen. But yeah, you can see the only thing he does is he went into send receive mode with phone C, right? And of course, phone C just 200 okay is that with the send receive. Yep, right, right there. And now we have two way path, right? Here's a, here's a two way path between them. We still have the two way path between A and B. And so now we have two simultaneous calls going on on two different lines on phone A. So what the phone is actually doing, it's just tying the media together behind the scenes. Fortunately, in this case, right here, we're all using, we all agreed on G seven 11 as our codec. However, if one of the phones like this first call was using G seven 29 and the second call was using G seven 11, then the phone would either have to transcode that media between the different calls, or it might try to renegotiate the RTP with, you know, one of the phones, you know, so like if this was if this car right here between A and C was in G seven 29, if this call was in G seven 29, he might try to up speed the whole thing with a with a reinvite to G seven 11 to match the call between A and B, right? That's if it can't transcode. And I doubt very seriously, if these phones could transcode it might be something to look at in the future, or people want to see that test, leave a comment and we'll set it up. But anyway, a lot of call agents will transcode. So if your PBX, let's say, you know, that's how you can get a lot of people on one conference call. And it doesn't matter where they come in, you know, if they're coming in G seven 11, G seven 29, or whatever, whatever type of codex using the, the PBX will transcode that audio so everyone can hear each other, right? So really the phone is doing all the work at this point and SIP has nothing to do with it. You know, it's not doing any kind of broadcast or multicast or any fun thing like that. The phone is doing all the grunt work. And so would the PBX. Okay. There you have it. And then of course, you know, when I hung up, sorry, he's right here. So when I hung up phone A, he immediately sent two buys out one to phone C and one to phone B. And we get a couple of tuner, okay, so that's the end of our call flow. All right. Don't forget you can support the network engineering video blog by donation using a credit card and PayPal or by purchasing products at the muxall store. Details and links are in the description under this video. Well, that's it for this video. 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