1 00:00:00,005 --> 00:00:03,003 - [Instructor] Voice over internet protocol or VoIP 2 00:00:03,003 --> 00:00:07,002 is a method to carry voice traffic digitally over a network. 3 00:00:07,002 --> 00:00:09,005 Initially, providers released services 4 00:00:09,005 --> 00:00:11,009 that emulated traditional phone systems, 5 00:00:11,009 --> 00:00:14,006 changing out analog lines for digital. 6 00:00:14,006 --> 00:00:17,009 More recently, closed services like Skype 7 00:00:17,009 --> 00:00:20,007 have started offering new alternatives. 8 00:00:20,007 --> 00:00:23,007 Session Initiation Protocol is a signaling 9 00:00:23,007 --> 00:00:26,003 and control protocol for communications. 10 00:00:26,003 --> 00:00:29,003 It is most commonly used for voice and video delivery 11 00:00:29,003 --> 00:00:33,009 over an IP network, but can also carry instant messaging. 12 00:00:33,009 --> 00:00:37,000 SIP is primarily responsible for establishing 13 00:00:37,000 --> 00:00:39,002 and terminating these communications, 14 00:00:39,002 --> 00:00:42,002 but not actually carrying voice traffic. 15 00:00:42,002 --> 00:00:47,000 SIP uses TCP and UDP port 5060 for plaintext traffic 16 00:00:47,000 --> 00:00:52,008 and uses TCP and UDP port 5061 for TLS encrypted traffic. 17 00:00:52,008 --> 00:00:56,004 SIP packets employ a structure very similar to HTTP, 18 00:00:56,004 --> 00:00:59,004 including header fields and encoding rules. 19 00:00:59,004 --> 00:01:03,000 SIP resources are marked with a uniform resource identifier, 20 00:01:03,000 --> 00:01:05,005 much like web resources. 21 00:01:05,005 --> 00:01:08,007 The URI scheme employed is usually in the form of 22 00:01:08,007 --> 00:01:13,003 sip:username:password@host:port. 23 00:01:13,003 --> 00:01:16,006 SIP will often be employed on a digital office phone, 24 00:01:16,006 --> 00:01:18,004 also known as a user agent, 25 00:01:18,004 --> 00:01:22,001 to terminate a local or remote SIP proxy server. 26 00:01:22,001 --> 00:01:25,004 A proxy acts both as a client and a server 27 00:01:25,004 --> 00:01:26,006 taking a call request 28 00:01:26,006 --> 00:01:29,004 and forwarding it to the nearest endpoint it knows. 29 00:01:29,004 --> 00:01:30,008 Not only will the call control 30 00:01:30,008 --> 00:01:34,002 between the handset and proxy server utilize SIP, 31 00:01:34,002 --> 00:01:37,001 but often various gateways will also. 32 00:01:37,001 --> 00:01:39,007 A gateway is a method to connect to other mediums 33 00:01:39,007 --> 00:01:41,001 like outside providers 34 00:01:41,001 --> 00:01:44,008 or the public switch telephone network or PSTN. 35 00:01:44,008 --> 00:01:48,007 H.232 is also a signal and control protocol 36 00:01:48,007 --> 00:01:50,008 for communication services. 37 00:01:50,008 --> 00:01:56,004 It commonly uses TCP and UDP ports from 1718 to 1720. 38 00:01:56,004 --> 00:01:58,005 H.232 was originally developed 39 00:01:58,005 --> 00:02:00,004 to facilitate teleconferencing, 40 00:02:00,004 --> 00:02:03,005 but was quickly adjusted to be more encompassing. 41 00:02:03,005 --> 00:02:06,005 H.323 is most often used today 42 00:02:06,005 --> 00:02:08,009 as a gateway for VoIP services. 43 00:02:08,009 --> 00:02:12,004 The Media Gateway Control Protocol is a plaintext signaling 44 00:02:12,004 --> 00:02:14,000 and control protocol. 45 00:02:14,000 --> 00:02:16,002 As the name implies, it acts as a gateway 46 00:02:16,002 --> 00:02:19,007 allowing a phone system to interface with a PSTN. 47 00:02:19,007 --> 00:02:22,005 Unlike H.323 gateways, 48 00:02:22,005 --> 00:02:25,003 MGCP gateways have very little intelligence 49 00:02:25,003 --> 00:02:27,000 as they act more as a conduit 50 00:02:27,000 --> 00:02:29,000 for the phone system to control. 51 00:02:29,000 --> 00:02:36,000 It utilizes TCP port 2428 and UDP port 2427. 52 00:02:36,000 --> 00:02:38,000 The Real-Time Transfer Protocol 53 00:02:38,000 --> 00:02:41,004 is what actually transfers voice and video traffic. 54 00:02:41,004 --> 00:02:44,009 RTP works in conjunction with the above signaling protocols 55 00:02:44,009 --> 00:02:48,007 as well as the RTP Control Protocol. 56 00:02:48,007 --> 00:02:52,001 RTCP periodically sends statistical information 57 00:02:52,001 --> 00:02:54,005 to participants in a media stream. 58 00:02:54,005 --> 00:02:56,006 It carries no voice traffic. 59 00:02:56,006 --> 00:02:59,000 It was originally developed in 1996, 60 00:02:59,000 --> 00:03:03,008 but was revised in 2003 as RFC 3550. 61 00:03:03,008 --> 00:03:06,006 RTP can detect out-of-sequence packets 62 00:03:06,006 --> 00:03:08,009 and can perform jitter correction. 63 00:03:08,009 --> 00:03:11,001 It also allows for multicast delivery 64 00:03:11,001 --> 00:03:13,009 to multiple destinations simultaneously. 65 00:03:13,009 --> 00:03:16,007 RTP employs an error concealment algorithm, 66 00:03:16,007 --> 00:03:19,002 which means it can sustain some packet loss 67 00:03:19,002 --> 00:03:20,009 without users noticing. 68 00:03:20,009 --> 00:03:23,001 Couple this with a need for audio packets 69 00:03:23,001 --> 00:03:25,006 to be transmitted as quickly as possible 70 00:03:25,006 --> 00:03:28,002 and it's clear why RTP employs UDP 71 00:03:28,002 --> 00:03:29,008 as its delivery mechanism. 72 00:03:29,008 --> 00:03:32,005 RTP has a broad range of ports assigned 73 00:03:32,005 --> 00:03:38,004 from 16387 to 32767 UDP. 74 00:03:38,004 --> 00:03:41,003 An RTP stream should be an even port number 75 00:03:41,003 --> 00:03:44,007 and the next odd port should be the RTCP port. 76 00:03:44,007 --> 00:03:47,007 The RTP profile type specifies the codecs 77 00:03:47,007 --> 00:03:49,005 used to encode the audio 78 00:03:49,005 --> 00:03:52,005 for the payload portion of the RTP packet. 79 00:03:52,005 --> 00:03:59,005 This can be methods such as G.711, G.723, G.726 and so on. 80 00:03:59,005 --> 00:04:02,000 There are a plethora of hosted SIP services 81 00:04:02,000 --> 00:04:03,002 out on the web now. 82 00:04:03,002 --> 00:04:06,001 For smaller organizations, nothing more than phones 83 00:04:06,001 --> 00:04:08,007 and an internet connection are necessary. 84 00:04:08,007 --> 00:04:10,001 For larger organizations, 85 00:04:10,001 --> 00:04:12,008 a single server can terminate SIP phones 86 00:04:12,008 --> 00:04:15,004 and also act as a voicemail server. 87 00:04:15,004 --> 00:04:18,003 Larger enterprises generally utilize multiple SIP 88 00:04:18,003 --> 00:04:19,009 and voicemail servers. 89 00:04:19,009 --> 00:04:22,009 Having VoIP experience is often a plus on a resume 90 00:04:22,009 --> 00:04:26,000 so don't hesitate to get your hands on some.