FundamentosFundamentos de de VozVozsobresobre el el protocoloprotocolo IP IP

((VoIPVoIP))

OBJETIVO:OBJETIVO:ComprenderComprender el el entornoentorno de de convergenciaconvergencia de de redesredes

de de vozvoz, , datosdatos y video y video queque se se estáestá llevandollevando a a cabocabo en en laslasredesredes de de telefoníatelefonía, , identificandoidentificando laslas tecnologíastecnologías existentesexistentesy y nuevosnuevos desarrollosdesarrollos queque puedenpueden llegarllegar a a establecerseestablecersecomocomo estándaresestándares parapara la la implementaciónimplementación de de redesredes de de VoiPVoiP

AhoraAhora, , veamosveamos lo lo quequesucedesucede en la en la

actualidadactualidad…………((ejemploejemplo de la de la compañiacompañia PingtelPingtel))

INTRODUCTIONINTRODUCTION

Antes de Antes de comenzarcomenzar esteeste cursocurso, , vamosvamos a a identificaridentificar el el entornoentorno en en queque se se estaesta

llevandollevando a a cabocabo la la convergenciaconvergencia de de vozvoz, , video y video y datosdatos..

((dibujedibuje un un esquemaesquema de de unauna red de red de telefoniatelefonia))

66

THE INTERESTTHE INTEREST

88

99

THE DEFINITIONTHE DEFINITION

1111

1212

1313

1414

1515

1616

>$200 a port

1717

Must support allStandard PSTNinterfaces andCompany interfaces!!

For analog voice

SS7 traslation to IP world

1818

VoIP carrier will look for

1919

A dial gateway’snumber

Gateway askfor account infoand party to call44 –ingland, 171 london,And send a setup to the gateway is closes to this destination

The destinationgateway receive it andlook for the called numberTo place a call in the PSTN

2020

2121

2222

Placing a gateway, the corporatecan save moneyIn calls betweenits offices by driven it by using IP

Recuperatethe initial upfront investments in lessthan a year

2323

the better quality service typicallyis going to come from a carrier whohas some control over theperformance of that network

2424

2525

2626

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

2727

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

2828

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

SERVICE EXAMPLESSERVICE EXAMPLES

3030

API: Aplication Programming Interface

3131

3232

3333

3434

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

3535

3636

THE ISSUESTHE ISSUES

3838

3939

INTERNET DEFINITIONINTERNET DEFINITION

4141

4242

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.

4343

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

4444

4545

RSVP was the leading prioritizationscheme, 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

4646

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

4747

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 thechance 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

4848

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

4949

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

THE THE QoSQoS

5151

What Is Quality of Service?What Is Quality of Service?QoSQoS refers to the ability of a network to provide better service torefers to the ability of a network to provide better service toselected network traffic over various underlying technologies selected network traffic over various underlying technologies including Frame Relay, Asynchronous Transfer Mode (ATM), including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and 802.1 networks, SONET, and IPEthernet and 802.1 networks, SONET, and IP--routed networks. In routed networks. In particular, particular, QoSQoS features provide better and more predictable features provide better and more predictable network service by: network service by: Supporting dedicated bandwidth Supporting dedicated bandwidth Improving loss characteristics Improving loss characteristics Avoiding and managing network congestion Avoiding and managing network congestion Shaping network traffic Shaping network traffic Setting traffic priorities across the network Setting traffic priorities across the network

VoIPVoIP NETWORK NETWORK PROTOCOLSPROTOCOLS

5353

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

5454

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

5555

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

5656

H.323 was designed first for supporting multimedia conferencing over a LAN

Let’s look at each element..

5757

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 our Layer 4 and our Layer 3 would be UDP, not TCP, because it's voice and IP.

5858

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.

5959

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

UDP for our transport.

6060

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

6161

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

6262

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 requiredfunctions, such as admission control.Whenever a terminal or a gateway wants to participate in a call, they must go to their gatekeeperand 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

6363

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

6464

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

6565

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.

6666

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 agatekeeper, 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

"If you get an incoming call toany of these telephone numbers,

send them to me at this IP addressbecause I can complete calls tothese destinations on the PSTN."

6767

6868

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."

6969

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.

7070

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;

7171

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.

7272

7373

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.

7474

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

7575

7676

7777

7878

7979

8080

8181

8282

8383

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

8484

8585

8686

8787

8888

8989

VoIPVoIP CALL EXAMPLECALL EXAMPLE

9191

9292

What's coming into the gateway is G.711PSTN digital speech. The first thing that wemust do in the gateway is compress that;

64 kb/s is just too much

9393

9494

9595

9696

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

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

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'vegot to come to the right balance

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

that

9797

9898

9999

100100

101101

102102

103103

SUMMARYSUMMARY

104104

The ENDThe END