1 1 00:00:00,000 --> 00:00:02,980 VoIP or Voice Over IP. 2 2 00:00:02,980 --> 00:00:05,080 Now, Voice Over IP is a system 3 3 00:00:05,080 --> 00:00:07,320 that digitizes your voice traffic 4 4 00:00:07,320 --> 00:00:10,310 so it can be treated like any other data on the network. 5 5 00:00:10,310 --> 00:00:11,610 You can do this by connecting it 6 6 00:00:11,610 --> 00:00:13,520 to what's known as an ATA device, 7 7 00:00:13,520 --> 00:00:16,200 which essentially is going to convert your analog voice 8 8 00:00:16,200 --> 00:00:17,820 to something digital that you can use 9 9 00:00:17,820 --> 00:00:19,180 and push over the network. 10 10 00:00:19,180 --> 00:00:20,470 So if you've already used Vonage 11 11 00:00:20,470 --> 00:00:21,680 and you had that physical device 12 12 00:00:21,680 --> 00:00:23,000 that you plugged your phone into 13 13 00:00:23,000 --> 00:00:24,670 and then connected that to your network, 14 14 00:00:24,670 --> 00:00:26,930 that is an ATA device. 15 15 00:00:26,930 --> 00:00:29,600 Now, you can use a fully digital environment as well. 16 16 00:00:29,600 --> 00:00:31,880 Something like Skype or Cisco phone 17 17 00:00:31,880 --> 00:00:34,780 that's connected over RJ45 or Cat5 cables 18 18 00:00:34,780 --> 00:00:37,170 and they use the power of Ethernet to go back 19 19 00:00:37,170 --> 00:00:39,790 to a call manager to make your phone calls. 20 20 00:00:39,790 --> 00:00:41,700 Either way, we're going to talk about both 21 21 00:00:41,700 --> 00:00:44,590 of these architectures a little bit more in this lesson. 22 22 00:00:44,590 --> 00:00:46,190 Now, when you're dealing with VoIP, 23 23 00:00:46,190 --> 00:00:48,230 it uses a protocol known as SIP, 24 24 00:00:48,230 --> 00:00:50,640 the Session Initiation Protocol. 25 25 00:00:50,640 --> 00:00:52,560 This is used to set up, maintain, 26 26 00:00:52,560 --> 00:00:54,670 and tear down each one of those calls 27 27 00:00:54,670 --> 00:00:58,100 and they give you that dial tone and maintain consistency. 28 28 00:00:58,100 --> 00:01:00,850 Now, VoIP is really popular because it can save your company 29 29 00:01:00,850 --> 00:01:03,690 a lot of money and provide enhanced services 30 30 00:01:03,690 --> 00:01:05,780 over a traditional phone system. 31 31 00:01:05,780 --> 00:01:07,820 When you talk about a traditional phone system, 32 32 00:01:07,820 --> 00:01:10,750 you'll usually hear this referred to as a PBX, 33 33 00:01:10,750 --> 00:01:12,860 which is a Private Branch Exchange, 34 34 00:01:12,860 --> 00:01:14,900 or you may even hear it called POTS, 35 35 00:01:14,900 --> 00:01:17,150 which is the Plain Old Telephone System. 36 36 00:01:17,150 --> 00:01:19,710 Now, the reason you get caller ID automatically, 37 37 00:01:19,710 --> 00:01:21,030 and then you can change names 38 38 00:01:21,030 --> 00:01:22,620 and you can even do video over it, 39 39 00:01:22,620 --> 00:01:24,920 it's all because you're using VoIP, 40 40 00:01:24,920 --> 00:01:26,800 which is a digital service. 41 41 00:01:26,800 --> 00:01:28,410 There are all sorts of things that you can do 42 42 00:01:28,410 --> 00:01:30,850 with a great VoIP solution that you just can't do 43 43 00:01:30,850 --> 00:01:33,660 on a private branch exchange or PBX. 44 44 00:01:33,660 --> 00:01:36,380 So, how does VoIP actually work? 45 45 00:01:36,380 --> 00:01:38,680 Well, you can run it inside a desktop computer 46 46 00:01:38,680 --> 00:01:40,930 in your browser, like using Google Voice, 47 47 00:01:40,930 --> 00:01:42,810 or you can use it with a handset, 48 48 00:01:42,810 --> 00:01:45,390 or you can see an IP phone that is dedicated 49 49 00:01:45,390 --> 00:01:46,860 like you see here on the screen. 50 50 00:01:46,860 --> 00:01:49,870 These are all different ways that you can use VoIP. 51 51 00:01:49,870 --> 00:01:51,460 Now, if I'm trying to talk to you 52 52 00:01:51,460 --> 00:01:54,390 from a digital saying like, Google Voice or Skype, 53 53 00:01:54,390 --> 00:01:56,920 and I want talk to your regular analog phone. 54 54 00:01:56,920 --> 00:01:58,700 Well, we're going to have to make some kind 55 55 00:01:58,700 --> 00:02:00,560 of transition between my digital world 56 56 00:02:00,560 --> 00:02:02,440 and your analog phone, right? 57 57 00:02:02,440 --> 00:02:03,890 So, how can we do that? 58 58 00:02:03,890 --> 00:02:06,620 Well, when I pick up the phone, an IP phone, 59 59 00:02:06,620 --> 00:02:09,080 it does a session initiation protocol 60 60 00:02:09,080 --> 00:02:11,430 down to what's called a Call Agent. 61 61 00:02:11,430 --> 00:02:14,330 That call agent then sets up the call through a router 62 62 00:02:14,330 --> 00:02:17,830 and it goes up and connects to another router or a gateway. 63 63 00:02:17,830 --> 00:02:20,960 That router or gateway is going to be tied to a PBX, 64 64 00:02:20,960 --> 00:02:22,700 a private branch exchange 65 65 00:02:22,700 --> 00:02:25,700 and this is going to initiate the call and that is going to create 66 66 00:02:25,700 --> 00:02:28,260 this session initiation protocol that we're using. 67 67 00:02:28,260 --> 00:02:30,050 And once that protocol is connected, 68 68 00:02:30,050 --> 00:02:33,230 we can then use RTP, the real-time protocol 69 69 00:02:33,230 --> 00:02:36,970 to actually pass the voice traffic during that session. 70 70 00:02:36,970 --> 00:02:40,040 Now, when we go from the IP phone, through the router, 71 71 00:02:40,040 --> 00:02:43,500 to the other router, and then to the PBX to make the call, 72 72 00:02:43,500 --> 00:02:45,760 that then goes into the analog system 73 73 00:02:45,760 --> 00:02:47,870 of the plain old telephone system. 74 74 00:02:47,870 --> 00:02:49,800 Now, if all of that is really confusing, 75 75 00:02:49,800 --> 00:02:52,610 don't worry too much about it because for the exam, 76 76 00:02:52,610 --> 00:02:55,140 you don't need to know how this typology works, 77 77 00:02:55,140 --> 00:02:58,110 except to know that VoIP is used to set up, 78 78 00:02:58,110 --> 00:03:01,510 maintain and terminate the call using SIP, 79 79 00:03:01,510 --> 00:03:03,550 the session initiation protocol. 80 80 00:03:03,550 --> 00:03:06,370 And once that call is set up and being maintained, 81 81 00:03:06,370 --> 00:03:08,100 the actual voice traffic goes through 82 82 00:03:08,100 --> 00:03:11,440 the real-time protocol or RTP. 83 83 00:03:11,440 --> 00:03:14,820 Now, next we have those PBX's and we can also create 84 84 00:03:14,820 --> 00:03:17,290 a virtual private branch exchange. 85 85 00:03:17,290 --> 00:03:19,140 Now, how do these relate to VoIP? 86 86 00:03:19,140 --> 00:03:20,900 Well, this allows you to have the ability 87 87 00:03:20,900 --> 00:03:22,700 to outsource your telephone system, 88 88 00:03:22,700 --> 00:03:25,520 because everything now is just ones and zeros. 89 89 00:03:25,520 --> 00:03:28,270 There's no reason that I can't just route this over to India 90 90 00:03:28,270 --> 00:03:30,390 and have them handle my PBX for me 91 91 00:03:30,390 --> 00:03:32,300 or some other cloud provider. 92 92 00:03:32,300 --> 00:03:34,230 Now, this is going to use VoIP to send 93 93 00:03:34,230 --> 00:03:36,230 all of your data to the provider, 94 94 00:03:36,230 --> 00:03:38,450 and they're going to connect it to whatever telephone system 95 95 00:03:38,450 --> 00:03:42,250 you need to use, whether that's US-based or overseas. 96 96 00:03:42,250 --> 00:03:45,490 The benefit of this is that you can have this virtual PBX 97 97 00:03:45,490 --> 00:03:48,040 that provide your voicemail, your caller ID, 98 98 00:03:48,040 --> 00:03:50,720 your messaging systems, and everything else for you. 99 99 00:03:50,720 --> 00:03:52,240 All you need to do is get and pay 100 100 00:03:52,240 --> 00:03:54,630 for that standard dial tone and then you're in business 101 101 00:03:54,630 --> 00:03:57,240 and you can start making phone calls all day long 102 102 00:03:57,240 --> 00:03:58,823 at a really inexpensive rate.