ICE L9 VoIP 004

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rnet Communication Engineering 23/07

yright RMIT University

Voice over IP

Mark A Gregory

RMIT University

10.8.6

0418 999 089

[email protected]

1 July 2006 Copyright RMIT Universi ty 2

Presentation Outline

Brief overview of VoIP and applications

Challenges of VoIP

IP Support for Voice

Protocols used for VoIP (current views)

RTP RTCP

RSVP H.323

SIP

1 July 2006 Copyright RMIT University 3

Objectives

Understand the current state of the art in

Internet Telephony.

Describe the key protocols involved in VoIP

and their roles in this new technology.

Discuss the limitations of current VoIP

technology.

Identify key factors influencing the growth of

Internet telephony services for globalcommunications.


References

Internet RFCs

RFC1889 - RTP

RFC 2205 - 2209 - RSVP

ITU-T

H.323 - Packet-based multimedia

communications systems

Voice over IP - Technology Guide Series

IEEE Network May/June 1999


Introduction

Many companies have seen advantages in

minimising costs by transporting voice over IP

networks.

This has set the stage for standards development

and the design of terminals and gateways and the

rolling out of services on a global scale.

Reliability

Quality


Introduction

Adding voice to packet networks

generates many challenges: interoperability

packet loss

delay

scalability

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Examples of Possible VoIP

Applications PSTN Gateways

PC based telephone accessing a public network by

calling a gateway at a point close to the destination tominimise long distance charges.

Internet aware telephones

Enhancement of ordinary telephones to serve as an

Internet access device as well as ordinary telephony.

Directory services could be accomplished via the

Internet.


1 July 2006 Copyright RMIT University 9 1 July 2006 Copyright RMIT Universi ty 10


Examples of Possible VoIP

Applications

Tie line replacement

Intranet links could replace tie lines betweencompany PBXs

Remote access from branch or home

Small office could gain access to corporatevoice, data and fax.

Voice calls from a mobile PC via theInternet

Internet call centres


More on Challenges

Voice quality has to be comparable to PSTN

Underlying network must meet strictperformance criteria including:

minimising call rejections

network latency

packet loss

disconnects

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More on Challenges

Call control (signalling) must make the

telephone calling process transparent so

that the callers need not know thetechnology involved.

PSTN / VoIP service interworking.

System management and security and

accounting and consolidated with PSTN

Operation Sub Systems


PSTNPSTN IPIP PSTNPSTNV ect r aOf f i ce

HEWLETTPACKARD

Telephone

Client

PSTN AccessGateway PC Client

Delay(End to End) Round trip

End to End Delay for VoIP PC

Phone Call

Above chart from IEEE Network May/June 1999

Author Bill Goodman Internet Telephony and Modem

Delay


IP Network Support for Voice

We need a network that is capable of

supporting real time telephone and fax.



Three techniques for improving networkQoS: Controlled networking environment

Capacity pre-planned and adequate performance

Management tools to configure network nodes, monitor performance,

and manage capacity and flow on a dynamic basis.

Control protocols and mechanisms Protocols such as RSVP (Resources Reservation

Protocol) and RTP (Real Time Protocol)



Real time voice traffic can be carried over IP

networks in 3 ways:

Voice trunks

Voice packets are transferred between pre-defined IP

addresses

Eliminate the need for phone number conversions.

Fall back to the PSTN as an option.

PC to PC Voice

Multimedia PCs can utilise this technique without connecting

to the PSTN.

System emulates an Internet Chat group and can be

combined with multimedia whiteboards.



Telephony

Appears like normal phone but employs various

forms of voice over packet networks.

Gateway functionality is required if the PSTN

needs to be accessed.

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IP Network Protocols currently being used

to implement VoIP:H.323

RTP, RSVP, RTCP

UDP/TCP

Network Layer (IPv4 and IPv6)

Data Link Layer

Physical Layer

We shall look at SIP, RTP, RTCP and RSVP to see theirfunctions. This will be followed by a brief overview of H.323.

SIP


RTP - Real-time Transport Protocol

Reference: RFC 1889

A real time end to end protocol

Utilises existing transport layers for data that

has real time properties.

Used by H.323

Takes the bitstream generated by the media

encoder breaking it into packets, sending the

packets over the network and recovering the

bitstream at the receiver.



Plays a key role in Internet telephony since it

is the component that moves the actual voice

among the participants.

Signalling protocols provide the parameters

for RTP transport.



Specific functions provided by RTP are:

Sequencing

Each RTP pack has a sequence number used

for loss detection and compensation for

reordering.

Intramedia synchronisation

Packets within the same stream can suffer

different delays (jitter). Applications use playout

buffers to compensate for this jitter. RTP

provides timestamps to assist in this.



Payload Identification

Since network conditions may vary during a call, it

may be necessary to change encoding

dynamically. RTP contains a payload type identifier

in each packet.

Frame Indication

Video and audio are sent in logical units called

frames. It is used to mark B of Frame and E of

Frame for upper layers.



Source Identification

In a multicast session, many users areparticipating and so there has to be a

mechanism for a packet to say which

participant actually sent it. A special identifier

called a SSRC - Synchronisation Source is

included in the protocol.

The RTP protocol has a companion

protocol called the Real Time Control

Protocol (RTCP).

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RTCP - Real Time Control Protocol

Identification

RTCP packets contain full details of email, phone

number and name of participant - this is availableto other participants.

Session Control

Allows you to send small notes or say goodbye!!



RFC 1889 defines five types of RTCP packets:

RR (Receiver Report)

SR (Sender Report)

SDES (Source Description Items)

BYE

APP



RR (Receiver Report): These are generated byparticipants that are not active senders. They containreception quality feedback about data delivery,including the highest packets number received, thenumber of packets lost, inter-arrival jitter, andtimestamps to calculate the round-trip delay betweenthe sender and the receiver.

SR (Sender Report): These are generated by activesenders. In addition to the reception quality feedbackas in RR, they contain a sender information section,providing information on inter-media synchronization,

cumulative packet counters, and number of bytessent.



SDES (Source Description Items): Contains information aboutthe sources.

BYE: Indicates that participation has ended.

APP: Intended for application specific purposes. It is onlyintended for experimental purposes.

RTCP control packets provide the following services: Quality of Service (QoS) monitoring and congestion control

Source identification

Intermedia synchronization

Control information scaling


RTSP

RTSP or Real-time Streaming Protocol is a client-servermultimedia presentation control protocol designed for

controlling streaming data over IP networks. It is definedin RFC 2326.

It was jointly developed by RealNetworks, NetscapeCommunications, and Columbia University. It waspublished in 1998 as a Proposed Standard by the IETF.

Streaming breaks data into many packets sizedappropriately for the bandwidth available between theclient and server.


RTSP

When the client has received enough packets,the user software can be playing one packet,

decompressing another and receiving the third.The user can begin listening immediately withoutthe need to download the entire file.

RTSP is an application-level protocol designedto work with lower level protocols such as RTPand RSVP to provide a complete streamingservice across IP networks.

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RTSP

RTSP provides VCR-style remote control functionalityfor audio and video streams, i.e., pause, fast forward,reverse, and absolute positioning. Sources of data can

include live data feeds or stored files.

Because of this RTSP is considered to be more of aframework rather than a protocol.

RTSP aims to be what HTTP is to textual data andgraphics data. However, there are key differencesbetween RTSP and HTTP:


RTSP

HTTP is a stateless protocol whilst RTSP isnt. The RTSPserver has to has to maintain session states in order tocorrelate RTSP requests with a stream.

HTTP is basically an asymmetric protocol, where the clientissues requests and the server responds. In RTSP, both theclient and server can issue requests.

Listed below are the methods to support the servicesand operations of RTSP: OPTIONS: The client or the server tells the other party the

options that it can accept.

DESCRIBE: The client retrieves the description of a presentationor media object identified by the request URL from the server.


RTSP

ANNOUNCE: When sent from client to server, ANNOUNCEposts the description of a presentation or media object identifiedby the request URL to a server. When sent from server to client,ANNOUNCE updates the session description in real-time.

SETUP: The client asks the server to allocate resources for astream and start an RTSP session.

PLAY: The client asks the server to start sending data on astream allocated via SETUP.

PAUSE: The client temporarily halts the stream delivery withoutfreeing server resources.

TEARDOWN: The client asks the server to stop delivery of thespecified stream and free resources associated with it.

GET_PARAMETER: Retrieves the value of a parameter of a

presentation or a stream specified in the URI.


RTSP

SET_PARAMETER: Sets the value of aparameter for a presentation or streamspecified by the URI.

REDIRECT: The server informs the clientsthat it must connect to another serverlocation. The mandatory location headerindicates the URL the client should connectto.

RECORD: The client initiates recording a

range of media data according to thepresentation description.


RSVP Protocol

Reference: RFC 2205 - 2209

General purpose signalling protocol thatallows network resources to be reserved

for a connectionless data stream, based

on receiver controlled requests.

Reserve

Reserve Reserve


RSVP

RSVP was jointly developed with Xerox

Corp.s Palo Alto Research Center(PARC), MIT and the Information Sciences

Institute of University of California (ISI).

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RSVP

RSVP sits on top of IP. However, it is not a routingprotocol but an internet control protocol. RSVP relies onthe underlying routing protocols to find where it should

deliver the reservation requests. RSVP works withunicast and multicast routing protocols.

The RSVP reservation process does not actuallytransmit the data nor does it provide the requested QoS.But it does guarantee that network resources areavailable when the transmission takes place. It shouldalso be noted that RSVP is just a general facility todistribute reservation parameters; it does not dictate howto set the connection parameters to achieve therequested QoS.


RSVP Data Flows

Filterspec

Flowspec QoS deli very

Best-effort

delivery

Packet Scheduler

Other packets

Packets of one session

(addressed to one destination)

Packets that pass

filter


RSVP Data Flows

Three concepts relating to data flows form the basis ofRSVP operation: Session

Flow specification

Filter specification

Session: A data flow identified by its destination. Once areservation is is made at a router by a particulardestination, the router considers this as a session andallocates resources for the life of that session.

A reservation request issued by a destination end

system is called a flow descriptor and consists of aflowspec and filterspec.


RSVP Data Flows

Flowspec: Specifies a desired QoS and is used

to set parameters in a nodes packet scheduler.

Filterspec: Defines the set of packets for which a

reservation is requested. Both the filterspec and

session define the set of packets, or f low that to

receive the desired QoS. Any other packets

addressed to the same destination are handled

as best-effort traffic.


Signalling

There are currently 3 major protocols that

support signalling in the IP network for VoIP

applications, viz:

H.323

MGCP

SIP

H.323 and SIP are described in the next few

slides.


Signalling

MGCP - Media Gateway Protocol

A control protocol allowing for the monitoringof events in IP phones and gateways and to

instruct them to send media to specific

addresses.

SIP - Session Initiation Protocol

Protocol developed by IETF for lightweight

call control and capabilities negotiation.

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H.323 Protocol

Reference: ITU-T H.323

Title: Packet-based MultimediaCommunications Systems

Conceived originally for multimedia

conferencing on a LAN, but now extended

to cover Internet Telephony. (Revised in

1998)


H.323 Protocol

Provides:

Call control

Conferencing functions

Call management

Capability negotiation

Supplementary services


H.323 Protocol

This Recommendation describes terminals

and other entities that provide multimedia

communications services over Packet Based

Networks (PBN) which may not provide a

guaranteed Quality of Service. H.323 entities

may provide real-time audio, video and/or

data communications.


H.323 Protocol

Support for audio is mandatory, while

data and video are optional, but if

supported, the ability to use a specified

common mode of operation is required,

so that all terminals supporting that

media type can interwork.


H.323 Protocol

The packet based network over which

H.323 entities communicate may be a point-to-point connection, a single

network segment, or an internetwork

having multiple segments with complex

topologies.


SIP

Reference: RFC3261

SIP provides the necessary protocol mechanisms so that

end systems and proxy servers can provide services:

call forwarding, including the equivalent of 700-, 800- and 900- type calls;

call-forwarding no answer;

call-forwarding busy;

call-forwarding unconditional;

other address-translation services;

callee and calling `number'' delivery, where numberscan be any (preferably unique) naming scheme;

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SIP

personal mobility, i.e., the ability to reach acalled party under a single, location-independentaddress even when the user changes terminals;

terminal-type negotiation and selection: a callercan be given a choice how to reach the party,e.g., via Internet telephony, mobile phone, ananswering service, etc.;

terminal capability negotiation;

caller and callee authentication;

blind and supervised call transfer;

invitations to multicast conferences.


SIP URI

SIP entities are identified using SIP URI

(Uniform Resource Identifier). A SIP URI has

form of sip:username@domain, for instance,sip:[email protected]. As we can see, SIP URI

consists of username part and domain name

part delimited by @ (at) character. SIP URIs are

similar to e-mail addresses, it is, for instance,

possible to use the same URI for e-mail and SIP

communication, such URIs are easy to

remember.


SIP

Extensions of SIP to allow third-party signaling (e.g., forclick-to-dial services, fully meshed conferences andconnections to multipoint control units (MCUs), as wellas mixed modes and the transition between those) areavailable.

SIP addresses users by an email-like address and re-uses some of the infrastructure of electronic mail deliverysuch as DNS MX records or using SMTP EXPN foraddress expansion. SIP addresses (URLs) can also beembedded in web pages. SIP is addressing-neutral, with

addresses expressed as URLs of various types such asSIP, H.323 or telephone (E.164).


SIP

SIP can also be used for signaling Internet real-time faxdelivery. This requires no major changes. Fax might becarried via RTP, TCP (e.g., the protocols discussed inthe Internet fax WG) or other mechanisms.

SIP is independent of the packet layer and only requiresan unreliable datagram service, as it provides its ownreliability mechanism. While SIP typically is used overUDP or TCP, it could, without technical changes, be runover IPX, or carrier pigeons, frame relay, ATM AAL5 orX.25, in rough order of desireability.


SIP Redirect Mode


SIP Proxy Mode

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Voice Gateway/Terminal

Functions The following picture shows the functional

components of terminals that use the

H.323 standards:

Voice

ProcessingCallProcessing

Network

Management

Packet

Processing

Speech

Signalling IPpackets

SNMPMessages


Other Associated Protocols

Among the many protocols relevant to

implementations of VoIP are:

TCP, UDP

IPv4 and IPv6

ATM and Frame Relay

SNMP

LDAP

WWW

etc


Software Support for VoIP

The software functionality required for

voice to packet conversion in a VoIP

gateway or terminal device is:

A Voice Processing module

Preparation of voice samples for transmission over

the packet network

A Call Processing module

Signalling gateway that allows calls to be

established across packet networks



A Packet Processing module

Processes voice and signalling packets by adding

the appropriate transport headers prior to

submitting the packets to the IP network.

Signalling information is converted from telephony

protocols to the packet signalling protocol.



A Network Management module

Management agent functionality

Remote fault

Accounting

Configuration management

Security?

Dialling directories

Remote access support.


Conclusions

This technology is in its infancy!

Many problems to overcome. Hot topics:

Quality of service

Echo cancellation

Manageability

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Conclusions

Many of the protocols that currently

support VoIP are not adequate for the task

and will need modification before they canbe really useful.

Need to determine tariff structures

also.

ICE L9 VoIP 004

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