Introduction to VoIP Part II

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Introduction to VoIPPart II

• Dr. Farid Farahmand
• CET479
• Updated 5/18/2007

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Overview

• Basic concepts of VoIP and its motivating facts
• How to digitally decode voice prior to its
  transport
• How to transport voice between users
• After the session is established how to transport
  voice
• How to setup and teardown voice sessions
• How to create sessions
• How signaling protocols work

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Speech Coding

• Voice has to be digitally encoded/decoded
• Streams of 1s and 0s
• How voice is coded impacts the channel efficiency
  (BW utilization)
• Various speech coding techniques are used
• Bandwidth and voice quality are related
• Yet the relation is not linear
• For example 16 Kbps voice transmission is not
  necessarily better than 32 Kbps
• Objective of speech coding is to minimize BW and
  maintaining high quality of speech
• High quality is measured by MOS metric
  (Mean-Option Score)
• Other metric alternatives are available (PSQM)

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A Little about Speech
More bits requires more BW but typically more
quantization level

• Speech is considered to be an analog signals
• The objective is to reconstruct the speech
  digitally

A signal can be reconstructed if the sampling
rate is twice the max. input frequency
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A Little about Speech

• Uniform quantization level can cause
  discrimination
• Loud voices will have lower quantization error
• A more effective approach is to us non-uniform
  quantization
• Smaller levels ? smaller quantization level
• Larger levels ? Less granularity

More accuracy
Less accuracy
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Speech-Coding Techniques

• Choice of speech coding is critical to having
  high-quality voice
• Two conflicting objectives
• Reducing bandwidth
• Maintaining the natural-sounding speech (toll
  quality)

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G. 711 Speech Coding

• ITU Recommendation G . 711 Speech decoding
• Typical human speech has a maximum frequency of
  about 4 KHz Fmax 4KHz
• Based on Nyquist Theorem, analog signals must be
  sampled at twice their maximum frequency
  Sampling rate 8000 sample/second 2 x Fmax
• Each sample is represented with 8 bits
• BW requirement will be 64 Kbps for standard
  circuit switch based telephone
• Toll-quality (MOS) is 4.3 Excellenet
• More efficient coding techniques
• G.726 ? 32 Bit rate (Kbps) toll-quality 4.0
• G.728 ? 16 Bit rate (Kbps) toll-quality 3.9
• G.729 ? 08 Bit rate (Kbps) toll-quality 4.0
• VoIP uses more efficient coding techniques
• The two ends negotiate on which coding technique
  to use

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Next

• Basic concepts of VoIP and its motivating facts
• How to digitally decode voice prior to its
  transport
• How to transport voice between users
• After the session is established how to transport
  voice
• How to setup and teardown voice sessions
• How to create sessions
• How signaling protocols work

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Transporting Voice Signals

• Digitally codes voice can be encapsulated into IP
  packets
• IP is just a routing protocol
• IP routing is based on the destination address
  packets with the same source/destination address
  can take different paths
• Provides no quarantine of service
• One way to transport the IP packet packets is
  using TCP
• The transmission control protocol (TCP)
• Ensuring that all packet are delivered in
  sequence
• Providing transmission reliability
• TCP provides port number in its header to
  distinguish between different applications (SMTP
  Port 25 / Web port 80 / Telnet Port 23)

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TCP/IP Model (Click for more information)
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TCP/IP Headers
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Introduction to UDP

• The User Defined Protocol performs a very simple
  function
• Passing IP packets to the end user
• Provides no guarantee of service and inherently
  unreliable
• Has no concept of packet ordering
• Yet, provides a quick one-shot transmission
• Most common example is using UDP in DNS

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UDP
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Voice over UDP

• UDP was not designed for transporting voice
• Due to its quick transporting ability, it is
  suitable for voice
• Basic shortcoming of UDP
• No packet loss recovery mechanism
• Voice communications can tolerate some loss
• Efficient coding techniques can be design to
  recover some lost packets
• Supporting QoS can reduce the probability of
  packet loss
• No packet ordering scheme
• Packets in the same session are unlikely to
  follow different paths ? lower probability of out
  of ordering

we still like to resolve some of the
shortcomings of UDP
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A Transport Protocol for Real-Time Application
Protocol (RTP)

• RTP is designed to support transporting real-time
  applications (voice, video, etc.)
• RTP contains two protocols
• RTP
• RTP Control Protocol
• Main functionalities
• Detect packet out-of-sequencing
• Report packet loss
• Only provides information and takes no action!

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RTP Protocols

• RTP resides on top of UDP
• Includes packet sequence number
• Provides timestamp (used for synchronization and
  calculating jitter and delay)
• RTP Control Protocol (RTCP)
• Considered as a companion to RTP / optional
• Provides feedback about quality of the voice
  session
• Number of lost RTP packets
• Packet delays
• Inter-arrival jitter
• RTP and RTCP are often established as two
  separate sessions
• Odd/Even port numbers between 1025-65,535

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Next

• Basic concepts of VoIP and its motivating facts
• How to digitally decode voice prior to its
  transport
• How to transport voice between users
• After the session is established how to transport
  voice
• How to setup and teardown voice sessions
• How to create sessions
• How signaling protocols work

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Call Setup and Teardown

• The main question
• How to establish a voice session
• How to teardown the session
• Call setup and teardown is commonly used in
  traditional telephony
• Signaling protocols are invoked before and during
  the call
• Setup
• Monitor/maintenance
• Teardown
• SS7 is the most common signaling example used in
  our telephone network
• In case of VoIP most initial signaling protocols
  were proprietary
• ITU-T (International Telecommunications Union
  Telecommunications Standardization Sector)
  recommended H.323 as the signaling protocol
• Version 1 1996
• Version 2 1998
• Version 4 Today!

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H.323 Architecture

• Basic components and scope
• Terminal
• Endpoints / end-user communication devices
• Multipoint control unit (MCU)
• An H.323 endpoint supporting multipoint
  conference
• Gatekeeper
• Optional entity
• Controls a number of H.323 terminal, gateways and
  MCUs
• Offers BW control services used to support QoS
• Gateway
• Establishes connection to other networks (etc.
  ISDN)
• Provides translation services between H.323 and
  other types of networks

• A set of terminals, MCUs, that a
• single gatekeeper controls is called a ZONE

SCN traditional switched circuit network (SCN)
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General Idea
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Overview of H.323 Protocols

• The actual signaling messages between H.323
  entities are specified by
• H.225 RAS Signaling
• H.223 Call Signaling
• H.245 Control Signaling
• H.225 has two parts
• Call Signaling
• The setup and teardown signaling is very similar
  to ISDN layer 3 spec. (Q.931)
• Can be carried over UDP or TCP / can be performed
  together whichever is established first
• RAS (registration, admission and Status)
  signaling
• Used between endpoint and a gatekeeper
• Always carried over UDP

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Overview of H.323 Protocols

• H.245 is a control protocol used between two or
  more endpoints
• Manages the media streams between H.323 session
  participants
• Establishes logical channels between endpoints
• The channel carries media streams between
  participants and include media type, bit rate,
  and so on

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References

• http//www.analog.com/library/analogDialogue/archi
  ves/40-04/blackfin_voip.html
• http//www.freesoft.org/CIE/RFC/1889/18.htm -
  RTCP