Fundamentos de Voz sobre el protocolo IP (VoIP)

Transcription

1 Fundamentos de Voz sobre el protocolo IP (VoIP)

2 OBJETIVO: Comprender el entorno de convergencia de redes de voz, datos y video que se está llevando a cabo en las redes de telefonía, identificando las tecnologías existentes y nuevos desarrollos que pueden llegar a establecerse como estándares para la implementación de redes de VoiP

3 Ahora, veamos lo que sucede en la actualidad (ejemplo de la compañia Pingtel)

4 INTRODUCTION

5 Antes de comenzar este curso, vamos a identificar el entorno en que se esta llevando a cabo la convergencia de voz, video y datos. (dibuje un esquema de una red de telefonia)

6 6

7 THE INTEREST

8 8

9 9

10 THE DEFINITION

11 11

12 12

13 13

14 14

15 15

16 >$200 a port 16

17 For analog voice Must support all Standard PSTN interfaces and Company interfaces!! SS7 traslation to IP world 17

18 VoIP carrier will look for 18

19 A dial gateway s number Gateway ask The destination for account info gateway receive it and and party to call look for the called number 44 ingland, 171 london, To place a call in the PSTN And send a setup to the gateway is closes to this destination 19

20 20

21 21

22 Recuperate the initial upfront investments in less than a year Placing a gateway, the corporate can save moneyin calls between its offices by driven it by using IP 22

23 the better quality service typically is going to come from a carrier who has some control over the performance of that network 23

24 24

25 25

26 for a Competitive Local Exchange Carrier and Incumbent Local Exchange Carrier might do as they begin to evolve to a hybrid situation where they've got their existing circuit switch network as well as the introduction of some IP telephony capabilities 26

27 a call that's not a local call but a toll call, a long distance domestic or international call, the voice switch would recognize that and throw this call into the IP switching domain 27

28 wireless carriers provide a service that, at least in the core of the network, is very similar to a wire line carrier 28

29 SERVICE EXAMPLES

30 API: Aplication Programming Interface 30

31 31

32 32

33 33

34 In an IP-based unified messaging system, I could call in, retrieve my messages, whether they're my fax messages, voice mail, or , and I can access them all from anywhere, wherever they happen to be and wherever I happen to be 34

35 35

36 36

37 THE ISSUES

38 38

39 39

40 INTERNET DEFINITION

41 41

42 IP is definitely a best-effort service; it does its best to deliver the packets--to forward the packets--to that ultimate destination. Sometimes it's successful, sometimes it's not just does its best. 42

43 IPv6, includes, in addition to the expanded address space, some capabilities to support security features; encryption, authentication Nevertheless, we have found ways to more efficiently allocate the IPv4 32-bit addresses, as well as we've figured out some tricks, like subnetting and dynamic address allocation, that has enabled us to extend the life of the 32-bit IPv4 address 43

44 44

45 RSVP was the leading prioritization scheme, but it has been surpassed by the protocol called DiffServ RSVP is still, at this point in the game, the prioritization scheme that's being recommended by H.323 version 2 for prioritizing your real-time voice traffic over the non-real-time data traffic in a Voice over IP environment 45

46 DiffServ requires that the routers in the network simply maintain multiple priority queues. So you have your normal priority and then your higher priority. But they're not required to actually actively manage the traffic in the network. That is done by the applications on the edge of the network 46

47 RSVP and DiffServ were prioritization schemes that required that we make changes to the routers throughout our IP networks to support these prioritization schemes. RTP instead provide some tools on the edge of the network to enable real-time applications to have the chance of achieving some better performance over that underlying non-real-time network. RTP does this by basically providing things like a time stamp, a sequence number, and even a performance monitoring mechanism. So what happens is RTP runs on the hosts on the edge of the network 47

48 RTCP sits on the receiving end and observes the performance of the RTP flow. Then periodically, RTCP will package up a performance report, send it back to the source, and the source will have the opportunity to use the information in that performance report to tweak the service --the application flow and the service provided to that 48

49 40 bytes of overhead before I get to the first bit of voice!! 49

50 THE QoS

51 What Is Quality of Service? QoS refers to the ability of a network to provide better service to selected network traffic over various underlying technologies including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and networks, SONET, and IP-routed networks. In particular, QoS features provide better and more predictable network service by: Supporting dedicated bandwidth Improving loss characteristics Avoiding and managing network congestion Shaping network traffic Setting traffic priorities across the network 51

52 VoIP NETWORK PROTOCOLS

53 the gateways in Voice over IP networks today uses the H.323 protocol 53

54 H.323 is actually a protocol that was designed to support multimedia conferencing over a LAN. when Voice over IP began to emerge, we needed something quick to get in and not miss the market opportunity. And H.323 seemed to be the best match at that time 54

55 H.323 is an overall protocol that points to a bunch of other protocols or other standards, 55

56 H.323 was designed first for supporting multimedia conferencing over a LAN Let s look at each element.. 56

57 audio codec supports at a minimum G.711. Then we'll also have, as a layer protocol, RTP, because we're generating real-time traffic here so we need our time stamps and our sequence numbers. And then, underlying, we've got our TCP/IP protocol stack--or our TCP and our IP--which in this case 57 our Layer 4 and our Layer 3 would be UDP, not TCP, because it's voice and IP.

58 Video, a real-time traffic type just like our voice, has a similar stack. For the video codecs we'll use at a minimum H.261; might also support H.263. Then we sequence our real-time packets with the Real-time Transport Protocol, RTP, and then UDP and IP. 58

59 For data we use, if data is supported, the T.120 standard. And then, because data doesn't have the same kind of real-time considerations that voice and video have, we don't have to use streamlined UDP for our transport. 59

60 The gateway is only required when we want to interface to the PSTN so when we want to speak to the outside PSTN world listed on the bottom there are all the various terminal types that are supported by an H.323 gateway. 60

61 The MCU is used as the central controller when we have a multimedia conference. 61

62 The gatekeeper is the overall manager of a portion of an H.323 network; we call that portion a zone. So, a gatekeeper controls an H.323 zone. And in controlling that zone, it provides a bunch of required functions, such as admission control. Whenever a terminal or a gateway wants to participate in a call, they must go to their gatekeeper and request permission. This is called admission control. another very important required function of that gatekeeper is what we call address translation. And address translation translates between a telephone number and an IP address 62

63 can provide some basic services similar to the kinds of vertical services that you see in the Public Switched Telephone Network often offered by a Service Control Point in the Intelligent Network. 63

64 we'll step through each stage of call setup and call processing. 64

65 the gateway is going to have to go and find an available zone that it can join. And it does that by sending a Gatekeeper Request message. The gatekeeper will come back with either a Gatekeeper Confirm or a Gatekeeper Reject. 65

66 "If you get an incoming call to any of these telephone numbers, send them to me at this IP address because I can complete calls to these destinations on the PSTN." next step is to join the gatekeeper s zone--to register--and we do that with a Registration Request message. Now notice that these are messages going between a gateway and a gatekeeper, and those messages are there for messages of the RAS protocol--the Registration/Admission/Status protocol--used to communicate between a gatekeeper and the nodes in its network 66

67 67

68 We use H.225 for contact the next gateway the first step is we send a Setup message. The gateway is going to send a Setup message--an H.225 Setup message--to that destination gateway that serves the user we're trying to call on the PSTN. Then that gateway--the destination gateway--is going to come back with a message we call call proceeding. And call proceeding basically says, "I'm proceeding with this call setup." Really what it does is buys you some time; "Reset your timers, give me some time I'm proceeding with the call setup, but it takes a little while." 68

69 Then that gateway--the destination gateway--is going to fall back into the RAS protocol, and it's going to go to its gatekeeper with an Admission Request, requesting permission to receive this incoming call. 69

70 The gatekeeper will confirm, granting permission for the call; and then at this point what we have is what we call an H.245 logical connection. H.245 is going to allow us to actually set up the media channels so we can begin to exchange media; 70

71 Capability Exchange is, we determine among the two endpoints, either terminals or gateways or gateways and terminals, what we can use to communicate with one another. You prefer G.729; okay, I'll accept G.729 from you. However, I prefer to use as my first choice, G.723 in the master/slave determination, what we do is determine who's going to be master, or controller, of that conference. 71

72 72

73 Logical channel number 0 is what we will use for the exchange of control information that's our control channel we have opened three additional channels: channel 4, 6, and 8, for audio, audio and video. We're ready to exchange our media at this point in time. 73

74 We've got our media payload, but we need to wrap it up in the Real-time Transport Protocol, which will provide those time stamps and those sequence numbers. Then we'll use UDP--the User Datagram Protocol--for our transport, our streamlined transport service 74

75 75

76 76

77 77

78 78

79 79

80 80

81 81

82 82

83 if the SIP server can be involved in the teardown of that call as well, or depending how we bill in this new environment 83

84 84

85 85

86 86

87 87

88 88

89 89

90 VoIP CALL EXAMPLE

91 91

92 What's coming into the gateway is G.711 PSTN digital speech. The first thing that we must do in the gateway is compress that; 64 kb/s is just too much 92

93 93

94 94

95 95

96 In fact, 60% of our frame here is the overhead of the protocol wrappings. That's a lot of overhead. Now we could add more speech and get a better overall bit efficiency here We could double the number of frames. We could have six frames of speech. That would give us 60 bytes of speech rather than 30 bytes of speech for the same 46 bytes of protocol overhead it doesn't come for free. If we're waiting around for the accumulation of three more frames of speech and the processing of that information, then we're wasting time. We're burning part of that delay budget, and that's delay that we can never recover from. So it's a trade off between your bit efficiency and your delay budget. You've got to come to the right balance 96

97 97

98 98

99 99

100 100

101 101

102 102

103 SUMMARY 103

104 The END 104