VoIP Overview

Voice Over IP

VoIP Overview

Instructor: Liibaan Cali Afrax

Email:[email protected]

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 VoIP Goals and Methods

• Original goal of VoIP is to deliver Telephone

Service as an IP Application over a Managed
IP Network. We will consider these three
items separately.

• Telephone Service
• IP Applications
• Managed IP Network

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 Telephone Service

• Telephone Service became a product over

100 years ago.
• A vast global network was designed to
support this service.
• The Public Switched Telephone Network
(PSTN) is optimized to carry telephone
service.

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 The meaning of Carrier Grade
• Important characteristics of Carrier Grade
Telephone service include:
– High Availability
– High capacity
– High quality of telephone service

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 Carrier Grade - Availability
• The network and its associated functions
and components is available 99.999 % of the
time

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 Carrier Grade - Capacity
• Millions of subscribers can place millions of
simultaneous calls

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 Carrier Grade - Telephone service
• calls are routed correctly
• calls are connected and disconnected within
a specified short period of time
• voice quality is high, as measured by
industry defined standards

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 Carrier Grade - Telephone functions
• We associate particular societal functions to
telephone service
• Emergency 911 service is the best example
• Government approved wiretapping is
another

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 Carrier Grade - Telephone features
• Traditional telephone networks identified
features and for the purpose of sales, they
grouped them into feature packages
• Features in these feature packages include:
• Call Waiting
• Call Forwarding
• Call Forward on Busy
• Message Waiting - LED indicator
• Message Waiting - audible indicator

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 IP Application
• The IP Protocol Suite places the Application
Layer at its top.
• VoIP is treated as an Application Layer
protocol similar to HTTP, FTP and SMTP.
• Most IP applications grew up after the
Internet Protocol and are optimized for use
on IP networks.
• Telephony grew up before the Internet
Protocol and the PSTN is optimized to
support telephone service.

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 IP Application Layer
• Applications such as
Telnet, HTTP and FTP Application
developed after [Telnet, SMTP, HTTP, FTP]

TCP/IP.
• VoIP attempts to
create telephone Transport
service using signaling [TCP, UDP, SCTP]

and media protocols

that operate at layer 5 Network

of the Internet Model Layer 2 - LLC

Layer 2 - MAC

Physical

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 Managed Data Network vs. The Internet

• Managed IP networks are different from the

Internet.
• The Internet is a collection of interconnected
networks.

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 Managed Data Network vs. The Internet
• No one manages the Internet.
• As a result it is not possible to predict the
path a packet of data will take or the time it
will take to traverse the network.
• Block data transfers may consume large
amounts of bandwidth at unpredictable times
and interfere with high priority - real-time
applications such as voice.

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 Managed Data Network
• A managed IP network has mechanisms in
place to control the allocation of bandwidth
based on routing and priority parameters.

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 VoIP Method

• Add specialized functional elements - VoIP

hardware and software - to the managed
network.
• The specialized elements create a media path
for packets of audio and video
• The specialized elements create a signaling
path for the call control commands :

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 Similar to SS7 system on PSTN

• This method has been used for many years on

the PSTN.
• The SS7 network is an overlay network that
carries signaling information.
• It adds specialized functional elements to the
PSTN network - STPs, SCPs and SPs.
• These elements use specialized protocols to
convey signaling information.

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 Signaling Path in the PSTN

Signaling Path

SS7
Voice Path
Overlay Network

Tandem
Switch
Telephone
Telephone

Transport Network
Access Network LE LE Access Network
Switch Switch Telephone
Telephone
Tandem
Switch

Telephone
Telephone

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 Signaling and Media on the IP Network

End Point

End Point
signaling path

End Point
media path
IP Network
End Point
signaling path

media path
End Point End Point

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 Signaling

• Signaling messages relate to the

management of media streams
• Signaling messages allow devices to
collaborate across the network to set up and
tear down bearer channels - streams of
audio and video information.

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 Signaling
• On the traditional PSTN, signaling
information includes:
– Ringing tones
– Dial tone
– Busy signals
– Congestion tones
– DTMF digits
– Dial pulse digits

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 VoIP Protocols
• In this course we study four VoIP protocols
specified by Internet Engineering Task Force
(IETF.)
– Three of these specifications are Signaling
Protocols:
• SDP - RFC 4566 (formerly RFC 2723)
• MGCP - RFC 3435
• SIP - RFC 3261
– One of them a Media Protocol:
• RTP - RFC 3550
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 IETF: Internet Engineering Task Force

• Access to IETF specifications and IETF

working groups is freely available to all.

• IETF Home Page : http://www.ietf.org/.

• RFCs : http://www.ietf.org/rfc.html.

• Working Groups :
http://www.ietf.org/html.charters/wg-dir.html.

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 IETF: Internet Engineering Task Force

• Working Groups focused on VoIP are found in the RAI

(Real-time Applications and Infrastructure) Area.
Included among them are:
– avt - Audio/Video transport
– bliss - Basic Level of Interoperability for SIP
Services
– ecrit - Emergency Context Resolution with Internet
Technologies
– enum - Telephone Number Mapping
– geopriv - Geographic Location/Privacy
– p2psip - Peer-to-Peer Session Initiation Protocol
– speermint - Session PEERing for Multimedia
INTerconnect

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 RTP and the Media Stream

• The Media Stream is encoded using a partial

protocol called Real-time Transport Protocol.
– RTP is carried in a transport layer protocol -
UDP
– RTP does not carry information about the
endpoints of the media stream
– RTP carries information about how the
media was encoded

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 RTP Packet

• The body of the RTP packet contains samples

of code that represent voice or some other
streaming media.

Header Sample 1 Sample 2 Sample 3 Sample n

• The header of the RTP packet provides

information about the method used to encode
the body.
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 RTP Header
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31
V=2 P X CC M PT Sequence Number 1
Timestamp 2
Synchronization Source (SSRC) ID 3
Contributing Source (CSRC) Identifiers ( 0 to 15 entries )

• PT - Payload Type - gives the name of the

CODEC.
• Sequence Number gives the order of the packet
in the stream.
• Timestamp enables the receiver to synchronize
playout of voice.

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 RTP Body

• The RTP payload is a digital representation of

streaming media - voice or video.
• A long list of CODECs (COder/DECoders) can be used
to encode the media.
• A list of the CODECs and associated parameters is
provided as a common resource for the internet
community by IANA the Internet Assigned Names
Authority.

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 Codecs
• There are three types of Codec:
– Waveform
– Vocoder
– Hybrid

• These are characterized by the:

– Voice Quality they produce
– Algorithmic delay they entail
– Bandwidth they use

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 Codec Reference

codec category type bandwidth in kbps comments

G.711 PCM wave-form 64 no algorithmic delay
G.721 ADPCM wave-form 32 no algorithmic delay

16; 24; 32; 40

G.726 ADPCM wave-form (supercedes G.721) no algorithmic delay
G.728 LD-CELP hybrid - AbS 16 algorithmic delay -> not good for voip
G.723.1 ACELP hybrid - AbS 6.3 or 5.3
G.729 ACELP hybrid - AbS 8
G.729A ACELP hybrid - AbS x sacrifice voice quality for speed of processing
G.729B ACELP hybrid - AbS x adds comfort noise and voice activity detection
G.729D ACELP hybrid - AbS 6.4
G.729E ACELP hybrid - AbS 11.8 robust in presence of background noise

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 Information in the IANA Codec list
PT encoding audio/video clock rate channels
name (A/V) (Hz) (audio)
_____________________________________________________________
0 PCMU A 8000 1
1 1016 A 8000 1
2 G721 A 8000 1
3 GSM A 8000 1
4 unassigned A 8000 1
5 DVI4 A 8000 1
6 DVI4 A 16000 1
7 LPC A 8000 1
8 PCMA A 8000 1
9 G722 A 8000 1
10 L16 A 44100 2
11 L16 A 44100 1
96--127 dynamic ?

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 Why Signaling Protocols?

• The body of the RTP packet contains

encoded voice.
• The header of the RTP packet contains
information that tells the receiver how to
decode the body.

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 Why Signaling Protocols?
• How do we create a stream of these packets?
• How do we connect that stream to the
processes that need to receive them?
• How do we alert the end-user that there is a
call?
• How do we disconnect the media stream from
the endpoints?

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 Why Signaling Protocols?

• Signaling Protocols enable us to do these

things.
• Signaling protocols carry the information you
usually expect to find in the header of a
protocol message.
• Signaling protocols are used instead of using
the header fields of the RTP.

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 VoIP Signaling Protocols

• IETF specifies two different VoIP signaling

protocols:
• Media Gateway Control Protocol - MGCP
• Session Initiation Protocol - SIP

• The two signaling protocols are distinguished

by an architectural choice - whether call control
is centralized or decentralized.

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 VoIP Signaling Protocols

• The Media Gateway Control Protocol

(MGCP) uses centralized call control.
• MGCP’s architecture is referred to as Smart
Core - Dumb Edge.
• The Session Initiation Protocol (SIP) uses a
decentralized call control.
• SIP’s architecture is referred to as Smart
Edge - Dumb Core.

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 MGCP Architecture
Blue line represents signaling path.
Red line represents talk path.

Network

Telephone
Call Agent Media Gateway

Media Gateway

Telephone

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 SIP Architecture

1 2 3 1 2 3
4 5 6 4 5 6
7 8 9 7 8 9
* 8 # * 8 #

SIP Proxy Server

SIP Registrar

Location
Service
1 2
4 5
3
6
SIP phone
7 8 9
* 8 #

1 2
4 5
3
6
SIP phone
7 8 9
* 8 #

Redirect Server
1 2 3
4 5 6
7 8 9
* 8 #

Managed IP Network

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 Signaling Message Structure
• The top line of the SIP and MGCP message together
with a number of header lines describe signaling
events such as ringing, ring-back, and message-
waiting.

• The body of the SIP and MGCP message contains

information about the IP address and port to which
the media will be delivered.

• The body of these messages is encoded in the fourth

VoIP protocol - Session Description Protocol - SDP.

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 Session Description Protocol
• The Session Description Protocol (SDP) is
used to carry the IP Address and UDP port
number of the endpoints of the RTP stream.
• In addition to this address and port information
the SDP carries other information such as…
– Mode - The session can be send-only,
receive-only or send/receive.
– Local Connection Options - The codec and
other details related to the encoding of the
voice.

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 Example SDP encoding

v=0
o=- 23456789 98765432 IN IP4
192.168.5.7
s=-
c=IN IP4 192.168.5.7
t=0 0
m=audio 6058 RTP/AVP 0
IP Address

UDP port Format Type

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 Example of MGCP Encoding

CRCX 1059 aaln/[email protected] mgcp 1.0

C: 9876543210abcdef
L: p:20, a:PCMU
M: recvonly

200 1059 OK
I: 456789fedcba5

v=0
o=- 23456789 98765432 IN IP4 192.168.5.7
s=-
c=IN IP4 192.168.5.7
t=0 0
m=audio 6058 RTP/AVP 0

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 SIP Invite Request and headers
Session Initiation Protocol
Request-Line: INVITE sip:[email protected]:5061 SIP/2.0
Message Header
From: "615551111"<sip:1 615551111 @bspgroup1.bsp.vonage.net:5061>;
tag=d82f8c0c-13c5-408a837c-1418ec-4591;user=phone
To: <sip:[email protected]:5061>;user=phone
Call-ID: 94529b7c-30968-1082819452-79026-128603335600569100000000-
[email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP 216.47.140.12:5061;branch=z9hG4bK-408a837c-1418ec-7cf6
User-Agent: <Motorola VT1000 mac: 000CE5C7AAB4 sw:VT20_1.1.16e ln:0
cfg:1082817753596>
Max-Forwards: 70
Contact: <sip:[email protected]:5061;transport=UDP;user=phone>
Content-Type: application/SDP
Content-Length:272

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 SIP Invite Request - body
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 16306144514 1082819452 1082819452
IN IP4 216.47.140.12
Session Name (s): SIP Call
Connection Information (c): IN IP4 216.47.140.12
Time Description, active time (t): 0 0
Media Description, name and address
(m): audio 10000 RTP/AVP 0 2 18 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:2 G726-32/8000
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): ptime:20

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 Web References
• IETF RFCs : http://www.ietf.org/rfc.html
• IETF working groups:
http://www.ietf.org/html.charters/wg-dir.html
• PacketCable Specifications:
http://www.packetcable.com/specifications/
• ETSI free standards download:
http://www.etsi.org/services_products/freestand
ard/home.htm

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 The End

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