presentation done in 2007

1. A glance at Voip
Nicola Marinelli

2. Agenda
 Classic Circuit Switched Telephony - PSTN
 Audio Signal - Sample-based Coding
 General Voip (why, advantages, QoS,...)
 H.323 (overview, elements, standard, …)

3. Classic Circuit Switched Telephony -
PSTN Public Switched Telephone Network
 POTS Plain Old Telephone Service -
Analog Telephony
 ISDN Integraded Services Digital Network -
Digital Telephony

4. Classic Circuit Switched Telephony -
POTS Plain Old Telephone Service
 Inventors
– Philipp Reise, Frankfurt 1861
– Alexander Graham Bell, Boston, 1876, patent
– Elisha Gray, 1876, 2 hours later
– Antonio Santi Giuseppe Meucci, 1854, presented in 1860, filed
patent in 1871, patent expired in 1874
 main usage (assumed): concert broadcasting
 voice coding and transmission: analog acoustic signal
transformed to (similar) analog electrical signal

5. Classic Circuit Switched Telephony -
POTS Plain Old Telephone Service
analog telephone connection (cont.):
 dedicated link between partners
 reserved bandwidth that cannot be used by other users =>
billing based on connection time and distance
 signaling (call setup and teardown):
– in the beginning: point-to-point links
– later telephone wired to central office

6. Classic Circuit Switched Telephony -
ISDN Integrated Service Digital Network
 transforming analog acoustic / electric signal to digital
values: PCM (Pulse Code Modulation)
8000 samples/s, each sample coded by 8 bits => 64 kbit/s

7. Classic Circuit Switched Telephony -
ISDN Integrated Service Digital Network
 PCM (Pulse Code Modulation)- Standard ITU-T G.711
– sampling: discrete time values
Nyquist’s Theorem: 1/Ts >= B for for a good quality signal conversion
analog => digital
Human voice: 1 sample every 125 million/s => 8000 sample/s
Shannon’s theorem: Fsampling >= 2* Fmax
Phone Signal (including the range of human voice) : 300 – 3400 Hz, it’s
considered 4000 Hz (Fmax) => sampling rate: 2* 4000 Hz => 8000 Hz
8 bits / sample value => 8*8000 =>64 kbit/s
– quantisation: discrete sampling values
splitting the voltage range chosen in intervals associating a single
digital value per each internal. 8 bits => 28
= 256 intervals

8. Classic Circuit Switched Telephony -
ISDN Integrated Service Digital Network
 An important step forward used in the ISDN technology
was the conversion of an analogue signal (human voice)
into a digital format, transporting it and, at the destination
party reconverting it again into the analogue format. This
coding approach is the one used in VOIP as well.
ISDN performs this anyway going on using mostly the
Circuit Switched Networks.

9. Classic Circuit Switched Telephony -
ISDN Integrated Service Digital Network
 BRI Basic Rate Interface (144 Kbps bandwidth)
1 D channel 16 Kbps for signaling
2 B channels 64 Kbps
 PRI Primary Rate Interface (2 Mbps bandwidth)
1 D channel 64 Kbps for signaling
30 B channels 64 Kbps

10. PSTN Public Switched Telephone Network
 Both POTS and ISDN use the same connection philosophy:
circuit switched, so there’s a permanent reservation of the
network resources for the active or on-going
communications.
 Both POTS and ISDN mostly use the same multiplexing
philosophy: TDM Time Division Multiplexing.
 Both POTS and ISDN use the same addressing standard:
E.164, so the classical telephone numbers.

11. VOIP Voice Over IP - A Definition
VOIP stands for Voice Over Intenet Protocol and is a
technology to let voice go through IP packets and,
in definitive through Internet so a packed switched
network.

12. VOIP Voice Over IP - An overview
To COde/DECode the voice, even Voip started using the PCM seen in
ISDN with further improvements. So, at the source site, an ADC (Analog
Digital Converter) makes the voice digital.The IP network transports
packets having an header to control the communication and payload to
transport data. At the destination site, another ADC, reconvert the digital
packet into the analog voice.

13. VOIP Voice Over IP - An overview

14. Why VOIP ? - Business Aspects
 single net infrastructure!! (companies, providers)
– Deutsche Telekom’s plan for 2012: PSTN will be totally replaced by
IP net
– USA: 30 % of telephone traffic is already VoIP
– the number of new VoIP providers is growing an growing monthly
 cheap costs!! (users)
– removal of recurring cost of leased PSTN lines and associated
equipments and their maintenance
– usage based (time/distance) charges substituted by flat
monthly/annual charges (ISP connection)
– less taxes

15. Why VOIP ? (cont.)
 universal VoIP account reachability!! (You can make a
call anywhere the Internet can reach)
 availability of new applications!!
– click-to-dial feature: link on a web page or application:
click => call setup
– unified messaging (telephony, e-mail, sms, chat..etc)
– exchange of additional data:
 audio and video conferences so multicast communication
 shared applications

16. QoS in VOIP
 VoIP goes through a network that was created to offer a best
effort service transport => queue time not controlled,
sometimes long:
– Packet loss => can be partially compensated by latest codecs
improving the voice compression phase and making shorter the voice
packet working on the headers (IP, UDP, RTP).
– Delay => calls may result in echo problems Jitter (delay variation)
compensated with a dejitter buffer => this buffer increases delay
when necessary.
 Security, all the same risks as can be found with any IP
technology.

17. QoS in VOIP- Packet loss
40 bytes only for headers!! For a transmission rate of 20ms at 8kbps there’s a payload (compressed voice)
of 20 kbps. So 66% of the packet is busy for headers with a large quantity of redundant data => so a
possible solution, only the header of the 1st packet is complete, the header of the following packets
only contain the delta => the whole header can became 4-5 bytes!

18. QoS in VOIP - delay
 The delay variation gives the jitter.
– optimal situation: no delay (=> no jitter):

19. QoS in VOIP - delay
 The delay variation gives the jitter effect.
– constant delay (=> no jitter):

20. QoS in VOIP - delay
 The delay variation gives the jitter effect.
– variable delay (=> jitter):

21. QoS in VOIP - delay
 The delay variation gives the jitter effect.
– jitter can be compensated by playout buffering
– example: buffering time long.
and risk of buffer overflow

22. QoS in VOIP - delay
 The delay variation gives the jitter effect.
– jitter can be compensated by playout buffering
– example: buffering time too short

23. QoS in VOIP - delay
 The delay variation gives the jitter effect.
– jitter can be compensated by playout buffering
– example: optimal buffering time

24. QoS in VOIP - security
 Security, all the same risks as can be found with
any IP technology
– typical solutions for data: IPSec, VPN
– encryption technology
At the end of March, Phillip Zimmerman, creator of the email
encryption technology PGP, announced that he had developed
Zfone, encryption for VoIP. It's currently only available on Mac
and Linux but Windows is coming.

25. QoS in VOIP - security
 Secure Real Time Protocol (SRTP)
Described in RFC 3711, provides confidentiality,
message authentication, and protection for RTP and
RTCP traffic.
As side effect, the packet creation needs more time.

26. Major Standards on VoIP
Minimal functions to be achieved:
– Signaling (sets up association between
parties)
 H.323
(ITU: International Telecommunication
Union)
 SIP (Session Initiation Protocol)
(IETF: Internet Eng. Task Force)
 Skype (company proprietary)
– Media transport (Voice and/or Video as
well)
 RTP & RTCP

27. Standards on VoIP - Media transport
Audio/Video(real-time) vs. Data
 Audio/Video:
– Analog in nature
– Delay sensitive
 Data:
– Loss sensitive
– Error sensitive

28. Standards on VoIP - Media transport
TCP vs. UDP
 TCP - Transmission Control Protocol
Connection oriented protocol.
– Handles sequencing and error detection,
ensuring that a reliable stream of data is
received by the destination application (packet
sequence numbers / acknowledgements /
resending).
– TCP – Signaling part with with upper layer
protocols.
 UDP - User Datagram Protocol
Connectionless protocol as IP
– Does not attempt to perform any sequencing, or
to ensure data reliability.
– UDP – Media Transport Part with upper layer
protocols.

29. Standards on VoIP - Media transport
RTP (Real-time Transport Protocol)
 provides end-to-end delivery services for data with real-time characteristics
(audio and video)
 Address real time requirements including Jitter(variation in delay times
experienced by the individual packets making up the data stream), buffering
and playing out at a constant rate.
 payload type identification=> allowing use of different coding schemes
 sequence numbering=>to discover packet loss and re-ordering
 time stamping=>allowing synchronization
 Source identification=>a source may be identified independently of its IP
address
 supports mixers and translators
 it’s used both in H.323 and SIP, major VOIP standards

30. Standards on VoIP - Media transport
RTP header format
T version P padding X extension CSRC ct. Sender counter
M Marker Payload Type media code method
Sequence Number seq. number of the packet for a media transmission
Timestamp timestamp for syncronization
SSRC identifier of the sender CSRC list of senders

31. Standards on VoIP - Media transport
RTP header format - translator/mixer features
 translator: coding transformation
 mixer: mixing several audio streams into a single audio stream

32. Standards on VoIP - Media transport
RTCP (RTP Control Protocol)
 RTCP functions:
– monitors and provides feedback on the quality of the transmission link for
data transported by the RTP packets (ie: if the bandwidth is lower than
usual, the sender may adapt the audio coding to a compression requiring
a lower data rate)
– support for multi-point communication(ie: information on leaving
conference members)
– identification of sources, Source DEScription:canonical name.(ie:for
communication conflict problems or program restart the SSRC identifier
might change, so the receiver asks for the CNAME (Canonical NAME) to
identify again the sender. It can be useful even to reorganize multiple
sessions (audio, video) with the same sender.

33. H.323 - a standard for VoIP
 Recommendation H.323 published by the International
Telecommunications Union Telecommunications Sector (ITU-T)
is a standard for multimedia services over packet networks,
including rules to map the multimedia. communication services
over a variety of networks.
 H.323 specifies:
– Components
– Protocols
– Procedures

34. H.323 - Components
 Terminals
 Multipoint Control Unit (MCU)
 Gateway
 Gatekeeper
 Border Elements
Referred to as
“endpoints”

35. H.323 - Components
Terminals
 An H.323 terminal is the communicating endpoint on the
network allowing the audio (and optionally even video)
bidirectional communication with another H.323 terminal,
gateway or MCU
Examples:
– Telephones
– Video phones
– IVR (Interactive Voice Responder) devices
– Voicemail Systems
– “Soft phones” (e.g.: NetMeeting®)

36. H.323 - Components
MCU (Multipoint Control Unit)
 Responsible for managing multipoint conferences
(two or more endpoints engaged in a conference)
 The MCU contains a Multipoint Controller (MC) that
manages the call signaling and may optionally have
Multipoint Processors (MPs) to handle media mixing,
switching, or other media processing

37. H.323 - Components
MCU (Multipoint Control Unit)
signaling and media data
signaling
media data

38. H.323 - Components
Gateway
 Gateways interface H.323 to other not H.323
networks mapping the transmission format and the
communication procedures
 The Gateway is composed of a :
– “Media Gateway Controller” (MGC) that handles call
signaling and other non-media-related functions
– “Media Gateway” (MG), that handles the media (multiplex,
rate matching, audio transcoding…)

39. H.323 - Components
Gateway
IP ISDN
Terminal
H.320
(ISDN)
Media Gateway Controller
signaling
Media Gateway
data
audio
video
Terminal
H.323
GatewayTerminal
H.323
Terminal
H.320

40. H.323 - Components
Gatekeeper
 Mandatory Zone Management functions :
– Admission control. Check if an endpoint is allowed or
not to access to the system.
– Address translation (routing). It works as a location
server mapping telephone number/email addresses/alias
to IP addresses.
– Bandwidth control and resource reservation

41. H.323 - Components
Gatekeeper - The Zone
 A Zone is a group of H.323 endpoints controlled by
one Gatekeeper
GK
GW
MCU
T1
Tn
PSTN
IP cloud
Router

42. H.323 - Components
Border Elements

43. H.323 - Protocols
Signaling (Everything that is not Media Data Transport)
 H.225.0 Call signaling protocol and media stream packetization for packet-based
multimedia. It includes Q.931 and RAS.
– H.225 RAS (Registration, Admission and Status)
 Gatekeeper discovery
 Gatekeeper Registration
 Admission control
– H.225 Call Signaling facility (derived from Q.931 ISDN control ch.)
 setup, alert, connect, release complete, ...
 H.245 provides “control” to the multimedia session that has been established - use
UDP
– Terminal capability exchange
– Master/Slave determinations
– Logical channel signaling
– Conference control (protocol for the MC channel in MCU)

44. H.323 - Protocols
Signaling
GK
Endpoint
Endpoint

45. H.323 Protocol Stack
IP
UDP
RTP
RTCP
TCP
Audio
Codecs
G.711
G.723.1
G.729
..
Video
Codecs
H.261
H.263
H.264
..
T.120
H.225.0
Call
Signaling
H.245
Control
Channel
H.225.0
RAS
Terminal Control and Management
Data
Applications
Media Control
Multimedia Applications, User Interface

46. H.323 - Procedures - basic call
Gatekeeper A Gatekeeper B
RRQ/RCF
ARQ
RRQ/RCF
LRQ
IP Network
Phone A
Gateway A Gateway B
H.225 (Q.931) Setup
H.225 (Q.931) Alert and Connect
H.245
RTP
ACF
LCF
V V
ARQ
ACF
Phone B

47. Thank You