introduction to voip part ii dr. farid farahmand cet479 updated 5/18/2007

Introduction to VoIPPart II

Dr. Farid Farahmand

CET479Updated 5/18/2007

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

Speech Coding

Voice has to be digitally encoded/decoded Streams of 1’s and 0’s

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)

A Little about Speech

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

Input Analog SignalHuman speech: 300-3800 Hz

Sampled Analog Signal Reconstructed Digital Signal

00010010001101010011…..

SamplingBlock

Quntization

Codec

0001

00100011

0101

SamplingFrequency

Input Analog Signal

A signal can be reconstructed if

the sampling rate is twice the max. input frequency

More bits requires more BW but typically more

quantization level

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

0001

00100011

0101

More accuracy

Less accuracy

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)

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

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)

TCP/IP Model (Click for more information)

The five layer TCP/IP model

5. Application layer

DHCP • DNS • FTP • HTTP • IMAP4 • IRC • NNTP • MIME • POP3 • SIP • SMTP • SNMP • SSH • TELNET • BGP • RPC • RTP • RTCP • TLS/SSL • SDP • SOAP • L2TP • PPTP • …

4. Transport layer

TCP • UDP • DCCP • SCTP • GTP • …

3. Network layer

IP (IPv4 • IPv6) • ARP • RARP • ICMP • IGMP • RSVP • IPSec • …

2. Data link layer

ATM • DTM • Ethernet • FDDI • Frame Relay • GPRS • PPP • …

1. Physical layer

Ethernet physical layer • ISDN • Modems • PLC • RS232 • SONET/SDH • G.709 • Wi-Fi • …

TCP/IP Headers

Input Analog SignalHuman speech: 300-3800 Hz

Input Analog Signal

Application layerDHCP • DNS • FTP • HTTP • IMAP4 •

IRC • NNTP • MIME • POP3 • SIP •SMTP • SNMP • SSH • TELNET •BGP • RPC • RTP • RTCP • TLS/SSL• SDP • SOAP • L2TP • PPTP • …

Transport layerTCP • UDP • DCCP • SCTP • GTP • …

Network layerIP (IPv4 • IPv6) • ARP • RARP • ICMP •

IGMP • RSVP • IPSec • …

Data link layerATM • DTM • Ethernet • FDDI • Frame

Relay • GPRS • PPP • …

Physical layerEthernet physical layer • ISDN • Modems

• PLC • RS232 • SONET/SDH • G.709• Wi-Fi • …

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

UDP

Field NameSize

(bytes)

Description

Source Port 2

Source Port: The 16-bit port number of the process that originated the UDP message on the source device. This will normally be an ephemeral (client) port number for a request sent by a client to a server, or a well-known/registered (server) port number for a reply sent by a server to a client. See the section describing port numbers for details.

Destination Port

2

Destination Port: The 16-bit port number of the process that is the ultimate intended recipient of the message on the destination device. This will usually be a well-known/registered (server) port number for a client request, or an ephemeral (client) port number for a server reply. Again, see the section describing port numbers for details.

Length 2Length: The length of the entire UDP datagram, including both header and Data

fields.

Checksum 2Checksum: An optional 16-bit checksum computed over the entire UDP datagram

plus a special “pseudo header” of fields. See below for more information.

Data Variable Data: The encapsulated higher-layer message to be sent.

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

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!

Data Link Layer

Physical Layer

Network Layer (IP)

UDP TCP

H.245Control

Signaling

H.225Call

Signaling

H.225RAS

SignalngRTCP

RTP

Voice/Video/Codec

Terminal / Application ControlAudio / VideoApplications

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

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!

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)

General Idea

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

Data Link Layer

Physical Layer

Network Layer (IP)

UDP TCP

H.245Control

Signaling

H.225Call

Signaling

H.225RAS

SignalngRTCP

RTP

Voice/Video/Codec


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

Data Link Layer

Physical Layer

Network Layer (IP)

UDP TCP

H.245Control

Signaling

H.225Call

Signaling

H.225RAS

SignalngRTCP

RTP

Voice/Video/Codec


References

http://www.analog.com/library/analogDialogue/archives/40-04/blackfin_voip.html

http://www.freesoft.org/CIE/RFC/1889/18.htm - RTCP

introduction to voip part ii dr. farid farahmand cet479 updated 5/18/2007

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