IP Telephony (VoIP)

1
IP Telephony (VoIP)

• CSI 4118
• November 30, 2004

2
Introduction (1)

• A recent application of Internet technology
  Voice over IP (VoIP) Transmission of voice over
  the Internet
• How VoIP works
• Periodically sample audio signal
• Convert each sample to digital form
• Send digitized stream across the Internet in
  packets
• Convert the stream back to analog for playback
• Why VoIP
• IP telephony is economic High costs for
  traditional telephone switching equipment
• Service flexibility

3
Introduction (2)

• Challenges
• Packet transmission delay (jitter)
• Call setup call establishment, call termination,
  etc.
• Backward compatibility with existing Public
  Switched Telephone Network (PSTN)
• IP Telephony Standards
• ITU (International Telecommunication Union)
  controls telephony standards
• IETF (Internet Engineering Task Force) controls
  TCP/IP standards

4
Encoding, Transmission, Playback (1)

• Both groups agree on the basics for encoding and
  transmission of audio
• Audio is encoded using a well-known technique,
  Pulse Code Modulation (PCM)
• The PCM audio is transferred using the Real-time
  Transport Protocol (RTP).
• UDP is used to transport the RTP messages
  encapsulated in a UDP datagram which in turn are
  encapsulated in IP Datagrams

5
PCM
Voice Band Width (BW) 4000 Hz (cycles per
second)
Sum of sine waves
v
v1
v2
v3
10010111 (one byte)
v4
t
125?sec
T1 Link speed 24 channels At each sample
time a frame is formed using one sample
from each channel 1 synch bit T1 rate
(24 X 8 1) / 0.000125 1.544 Mb/s
6
Encoding, Transmission, Playback (2)

• UDP is used for transport because
• lower overhead than TCP audio can be played
  sooner
• Playback cannot wait for a retransmissions
• Two independent RTP sessions exist, because an IP
  phone call involves transfer in two directions
• IP phone acts as sender for outgoing data, and
• IP phone acts as receiver for incoming data

7
RTP

• Problems in IP networks
• a correct audio packet, but delayed 1 minute is
  useless
• an out-of-sequence audio packet is useless
• How problems arise
• IP packets wait in router queues
• If router load is heavy, the packet is delayed
• If router has no load, the packet is processed
  immediately
• The delay difference is called jitter
• Packets take different routes and can arrive
  out-of-sequence
• RTP reduces jitter
• packets are time stamped and sequence numbered
• packets are ordered in a queue and played
  according to time stamp
• some other sound is played in place of discarded
  packet

8
A Basic IP Telephone System

• The simplest IP telephone system uses two basic
  components
• IP telephone end device allowing humans to
  place and receive calls
• Media Gateway Controller providing overall
  control and coordination between IP phones
  allowing caller to locate callee (e.g. call
  forwarding)

9
Interconnection with Others (1)

• IP telephone system needs to interoperate with
  PSTN or another IP telephone system
• Additional components needed for such
  interconnection
• Media Gateway
• Signaling Gateway
• Gateway Controller

10
Interconnection with Others (2)

• Media gateway translates audio between IP
  network and PSTN
• Signaling Gateway translates signaling operations

11
Signaling Systems Protocols

• Main complexity of VoIP Call setup and call
  management
• The process of establishing and terminating a
  call is called Signaling
• In traditional telephone system, signaling
  protocol is SS7 (Signaling System 7)
• In VoIP, signaling protocols are
• Recommendation H.323, by ITU
• Megaco MGCP, jointly by IETF and IUT.
• SIP (Session Initiation Protocol), by IETF
• VoIP signaling protocols should be able to
  interact with SS7

12
SS7 for PSTN

• SS7 is the most widely used Common Channel
  Signalling (CCS) protocol
• also known as out-of-band signalling
• No modern PSTN can function without SS7
• SS7 requires additional network elements
• Signal Transfer Point (STP)
• Service Control Point (SCP)
• STP is made very reliable by redundancy
• mated pairs separated by a distance
• Switches, or Service Switching Points (SSP) are
  very expensive and are never made redundant
• SS7 is fundamental to Intelligent Networks

13
SS7 PSTN Network
SSP - Service Switching Point - Switch with SS7
capabilities SCP - Service Control Point -
Database performing translations STP - Signal
Transfer Point - Packet-switching of signalling
information

• Reliability
• mated STP pairs
• redundant links

14
Out-of-band Signalling
15
ITU Recommendation H.323Packet-based Multimedia
NetworkingReal-Time Audio, Video, Data
Supports multipoint conferencing between
Terminals and Gateways
MCU (Mulitpoint Control Unit)
Terminal
Network
Gatekeeper
Gateway
Terminal
Terminal
Controls access to services of a Gateway
Discover appropriate Gatekeeper, register, then
establish sessions with other Terminals
Other networks PSTN, ATM, Internet
16
H.323 is a Signaling Protocol

• H.323, standardized by ITU, defines four
  elements
• Terminal IP phone
• Gatekeeper provides location and signaling
  functions coordinates operation of Gateway.
• Gateway used to interconnect IP telephone system
  with PSTN, handling both signaling and media
  translation.
• Multipoint Control Unit provides services such
  as multipoint conferencing.

17
H.323 Characteristics

• H.323 consists of a set of protocols that work
  together to handle all aspects of communication,
  including
• Transmission of a digital audio phone call
• Signaling to set up and manage phone call
• Allows transmission of video and data while a
  phone call is in progress
• Sends binary message
• Incorporates protocols for security
• Uses a special hardware Multipoint Control Unit
  for conferencing calls
• Defines servers for address resolution,
  authentication, accounting, features, etc.

18
H.323 Layering

• H.323 uses both UDP and TCP over IP.
• Audio travels over UDP
• Data travels over TCP

19
MGCPOver UDP with subscribe and packet
retransmission mechanisms
IP Side
PSTN Side
With translation to SIP
End Point (Gateway - signalling)
PSTN Signalling
VoIP Signalling
MGCP Phones
Analog Phones
MGCP
Call Agent (CA) (Media Gateway Controller)
CA issues commands to several types of
Gateways Gateways act on these commands
MGCP
End Point (Gateway - media)
Voice Signals
VoIP Packets
Converts PSTN media to VoIP and vice versa
20
MEGACOYet another Gateway Control Protocol

• MEGACO concepts
• Termination (T)
• entities such as a link, channel, individual
  party
• Context (C)
• collection of Terminations
• Event
• condition such as off-hook, on-hook, collection
  of dialed digits
• Packages
• Termination profiles - properties, events,
  signals, statistics, IDs
• MEGACO defines
• how to add, subtract and move Terminations
  between Contexts
• support for operations from simple telephone
  access to interfaces to different networks

21
MEGACO MG and MGC
22
MEGACO Architecture
23
IP Signaling Protocol

• SIP Session Initiation Protocol by IETF
• SIP defines three main elements that comprise a
  signaling system
• User Agent IP phone or applications
• Location servers stores information about users
  location or IP address
• Proxy and support servers
• Proxy Server forwards requests from user agents
  to another location.
• Redirect Server provides an alternate called
  partys location for the user agent to contact.
• Registrar Server receives users registration
  requests and updates the database that location
  server consults.

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SIP Network in SDL
25
SIP Call
26
What can be SIP-enabled
27
Registration
Location server
I am John Smith. I will be be reachable At
sipJohn.Smith_at_131.160.1.112
2
1
Registrar
I am John Smith. I will be be reachable At
sipJohn.Smith_at_131.160.1.112
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Proxy SIP Server
Proxy Server
(1) Invitation to a session for sipJohn.Smith_at_com
pany.com
(2) Invitation for a session for John.Smith_at_
131.160.1.112
131.160.1.112
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Redirect Server
131.160.1.112
(3) Invitation to a session for sipJohn.Smith_at_131
.160.1.112
(2) He is at is at sipJohn.Smith_at_131.160.1.112
SIP Server
(1) Invitation to a session for sipJohn.Smith_at_com
pany.com
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The SIP Trapezoid RFC 3261
atlanta.com . . .
biloxi.com . proxy
proxy . .
. Alice's . . . . .
. . . . . . . . . . . . . . . Bob's
softphone
SIP Phone
INVITE F1

---------------gt INVITE F2
100 Trying F3 ---------------gt
INVITE F4 lt--------------- 100
Trying F5 ---------------gt
lt-------------- 180 Ringing F6
180 Ringing F7
lt--------------- 180 Ringing F8
lt--------------- 200 OK F9
lt--------------- 200 OK F10
lt--------------- 200 OK F11
lt---------------
lt---------------
ACK F12
-------------------------
------------------------gt
Media Session
lt
gt BYE F13
lt------------------------
-------------------------
200 OK F14
-------------------------------------------------
gt
 
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INVITE F1
F1 INVITE Alice -gt atlanta.com proxy   INVITE
sipbob_at_biloxi.com SIP/2.0 Via SIP/2.0/UDP
pc33.atlanta.com Max-Forwards 70 To Bob
ltsipbob_at_biloxi.comgt From Alice
ltsipalice_at_atlanta.comgt Call-ID
a84b4c76e66710 CSeq 314159 INVITE Contact
ltsipalice_at_pc33.atlanta.comgt Content-Type
application/sdp Content-Length 142
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F2 INVITE
F2 INVITE atlanta.com proxy -gt biloxi.com
proxy   INVITE sipbob_at_biloxi.com SIP/2.0 Via
SIP/2.0/UDP bigbox3.site3.atlanta.com Via
SIP/2.0/UDP pc33.atlanta.com Max-Forwards 69 To
Bob ltsipbob_at_biloxi.comgt From Alice
ltsipalice_at_atlanta.comgt Call-ID
a84b4c76e66710 CSeq 314159 INVITE Contact
ltsipalice_at_pc33.atlanta.comgt Content-Type
application/sdp Content-Length 142
33
F3 100 Trying
F3 100 Trying atlanta.com proxy -gt
Alice   SIP/2.0 100 Trying Via SIP/2.0/UDP
pc33.atlanta.com To Bob ltsipbob_at_biloxi.comgt From
Alice ltsipalice_at_atlanta.comgt Call-ID
a84b4c76e66710 CSeq 314159 INVITE Content-Length
0
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F4 INVITE
F4 INVITE biloxi.com proxy -gt Bob   INVITE
sipbob_at_192.0.2.4 SIP/2.0 Via SIP/2.0/UDP
server10.biloxi.com Via SIP/2.0/UDP
bigbox3.site3.atlanta.com Via SIP/2.0/UDP
pc33.atlanta.com Max-Forwards 68 To Bob
ltsipbob_at_biloxi.comgt From Alice
ltsipalice_at_atlanta.comgt Call-ID
a84b4c76e66710 CSeq 314159 INVITE Contact
ltsipalice_at_pc33.atlanta.comgt Content-Type
application/sdp Content-Length 142
35
F5 100 Trying
F5 100 Trying biloxi.com proxy -gt atlanta.com
proxy   SIP/2.0 100 Trying Via SIP/2.0/UDP
bigbox3.site3.atlanta.com Via SIP/2.0/UDP
pc33.atlanta.com To Bob ltsipbob_at_biloxi.comgt From
Alice ltsipalice_at_atlanta.comgt Call-ID
a84b4c76e66710 CSeq 314159 INVITE Content-Length
0
36
SIP Characteristics

• Operates at the application layer.
• Encompasses all aspects of signaling, e.g.
  location of called party, ringing a phone,
  accepting a call, and terminating a call.
• Provides services such as call forwarding
• Relies on multicast for conference calls
• Allows two sides to negotiate capabilities and
  choose the media and parameters to be used
• SIP URI is similar to email address. (with prefix
  sip) E.g. sipbob_at_somewhere.com

37
SIP Methods

• Six basic message types, known as methods

38
An Example SIP Session

• User agent A contacts DNS server to map domain
  name in SIP request to IP address
• User agent A sends a INVITE message to proxy
  server that uses location server to find the
  location of user agent B
• Call is established between A and B. Then media
  session begins
• Finally, B terminates the call by sending a BYE
  request

39
Telephone Number Mapping Routing (1)

• How should users be named?
• PSTN follows ITU standard E.164 for phone
  numbers. E.g. 1-613-123-4567
• SIP uses IP addresses. E.g. sipsmith_at_uottawa.ca
• In an integrated network (PSTN IP), two
  problems defined
• Locate a user
• Find an efficient route to the user
• IETF proposed two protocols
• ENUM E.164 NUMbers
• TRIP Telephone Routing over IP

40
Telephone Number Mapping Routing (2)

• ENUM
• Converting E.164 phone number into a Uniform
  Resource Identifier (URI)
• Using Domain Name System to store mapping
• A phone number is converted into a special domain
  name e164.arpa
• E.g. 1-800-555-1234 ? 4.3.2.1.5.5.5.0.0.8.e164.arp
  a

41
Telephone Number Mapping Routing (3)

• TRIP
• Finding a user in an integrated network
• Used by location server or other NEs to advertise
  routes
• Independent of signaling protocols
• Dividing the world into a set of IP Telephone
  Administrative Domains (ITADs)

42
IP Telephones and Electrical Power

• Analog telephone system continues to work when
  electrical power are unavailable
• The wires that connect a telephone to the central
  office supply the power
• Currently, IP telephones have to depend on an
  external source of power
• IP phones must have both network connection and
  power connection.
• Several mechanism proposed to integrate power
  with network connections.

43
Summary (1)

• IP telephony or VoIP refers to the transmission
  of voice telephone calls over IP networks.
• Hot area both in research and market because of
  low cost
• Challenge in backward compatibility with PSTN
• The complexity of IP telephony is on signaling.
  Both ITU and IETF propose signaling standards.
• H.323, by IUT
• SIP, by IETF, offering similar functions to
  H.323, but simpler than H.323.
• Both are competing to be recognized as 1
  signaling protocol

44
Summary (2)

• H.323 uses a set of protocols for call setup and
  management
• SIP uses a set of servers to handle various
  aspects of signaling
• ENUM maps an E.164 telephone number into a URI
  (usually SIP URI)
• TRIP provides routing among IP telephone
  administrative domains
• IP telephones depends on external power, while
  analog phones dont.