Fundamentos de Voz sobre el protocolo IP \(VoIP\)

Fundamentos de Voz
sobre el protocolo IP
(VoIP)
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
Ahora, veamos lo que
sucede en la
actualidad……(ejemplo de la
compañia Pingtel)
INTRODUCTION
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
THE INTEREST
8
9
THE DEFINITION
11
12
13
14
15
>$200 a port
16
For analog voice
Must support all
Standard PSTN
interfaces and
Company interfaces!!
SS7 traslation to IP world
17
VoIP carrier will look for
18
A dial gateway’s
number
The destination
Gateway ask
gateway receive it and
for account info
look for the called number
and party to call
To
place a call in the PSTN
44 –ingland, 171 london,
And send a setup to the gateway is closes to this destination
19
20
21
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
the better quality service typically
is going to come from a carrier who
has some control over the
performance of
that network
23
24
25
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
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
wireless carriers provide a service that, at least in the core of the network, is very similar to a wire line carrier
28
SERVICE EXAMPLES
API: Aplication Programming Interface
30
31
32
33
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
34
35
36
THE ISSUES
38
39
INTERNET DEFINITION
41
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
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
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
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
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
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
40 bytes of overhead before I get to the first bit of voice!!
49
THE QoS
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 802.1 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
VoIP NETWORK
PROTOCOLS
the gateways in Voice over IP networks today uses the H.323 protocol
53
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
H.323 is an overall protocol that points to a bunch of other protocols … or other standards,
55
H.323 was designed first for supporting multimedia conferencing over a LAN
Let’s look at each element..
56
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
57our
Layer 4 and our Layer 3 would be UDP, not TCP, because it's voice and IP.
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
58
Real-time Transport Protocol, RTP, and then UDP and IP.
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
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
The MCU is used as the central controller when we have a multimedia conference.
61
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
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
we'll step through each stage of call setup and call processing.
64
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
"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
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
68
the call setup, but it takes a little while."
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
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
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
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
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
76
77
78
79
80
81
82
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
85
86
87
88
89
VoIP CALL EXAMPLE
91
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
94
95
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
We could double the number of frames. We could have six frames of speech. that delay budget, and that's delay that we can
That would give us 60 bytes of speech rather than 30 bytes of speech for the never recover from. So it's a trade off between
same 46 bytes of protocol overhead
your bit efficiency and your delay budget. You've
got to come to the right balance
96
97
98
99
100
101
102
SUMMARY
103
The END
104