Voice Over Internet Protocol (VoIP)

CSC550 Term Project

Group Members:Alice Miller, Bill Smith and Cathy Davis

Advisor: Dr. Frank Lee10/8/2005

OUTLINEOUTLINEINTRODUCTION

ADVANTAGES OF VoIP

POPULAR VoIP PROTOCOLS

H.323

SIP

MGCP

SUPPORTING PROTOCOLS

TECHNICAL ISSUES

HARWARE REQUIREMENTS

SOFTWARE REQUIREMENTS

PRODUCTS

SERVICES

FUTURE DEVELOPMENTS

CONCLUSION

INTRODUCTIONINTRODUCTION

VoIP - The ability to carry toll quality voice using compression

techniques and packet switching over the IP packet network.

Voice

CODEC:

Analog to Digital

Compress

Create Voice Datagram

Add Header

(RTP, UDP, IP etc)

CODEC:

Digital to Analog

Decompress

Re-Sequence and

Buffer-Delay

Process Header

Voice

analog analog

digital digital

INTRODUCTION (cont’d…)INTRODUCTION (cont’d…)

Real time voice traffic can be carried over IP

networks in three different ways1. PC to PC

2. PC to Phone

3. Phone to Phone

INTRODUCTION (cont’d…)INTRODUCTION (cont’d…)

Protocols commonly implemented

by Voice over IP

1. H.323

2. SIP (Session Initiation Protocol)

3. MGCP (Media Gateway Control

Protocol)

4. RSVP (Resource Reservation

Protocol)

ADVANTAGES OF VoIPADVANTAGES OF VoIP

INTEGRATION OF VOICE AND DATA: Web servers capable of

interacting with voice, data and images.

SIMPLIFICATION: Allows more standardization and less

equipment management.

NETWORK EFFICIENCY: Provides bandwidth consolidation.

COST REDUCTION: Slashes high charges for long distance calls.

ADVANCED APPLICATIONS: To be derived from multimedia and

multi-service applications.

H.323H.323

H.323 is a standard that specifies the

components, protocols and procedures

that provide multimedia communication

services such as real-time audio, video,

and data communications over packet

networks, including Internet Protocol (IP)

based networks.

COMPONENTS OF H.323

1. TERMINALS : Can be either a personal computer or

a stand-alone device

2. GATEWAYS : A H.323 gateway provides

connectivity between an H.323 network and a non-

H.323 network.

3. GATEKEEPERS : Provide call control services such

as address translation, bandwidth management,

admission control and zone management.

4. MULTIPOINT CONTROL UNITS (MCU) : Manage

conference resources, negotiate between

terminals for the purpose of determining the

audio or video coder/decoder to use, and may

handle the media stream.

Layout of H.323-enabled inter-network

H.323 PROTOCOL ARCHITECTUREH.323 PROTOCOL ARCHITECTURE

An integrated set of software programs that follows the

ITU (Int’l Telecomm Union) H.323 recommendation and

all associated recommendations.

CALL CONTROL LAYER

1. Signaling for call setup and capability exchange

2. Signaling of commands, indications and messages

to open

3. Describes the content of logical channels.

4. Formats the data streams into messages for output

5. Performs logical framing, sequence numbering, and

error detection and correction for each media type.

H.323 PROTOCOL ARCHITECTURE (cont’d)H.323 PROTOCOL ARCHITECTURE (cont’d)

CALL SIGNALING

1. The H.225 standard defines a layer that

formats the transmitted video, audio, data,

and control streams for output to the network,

and retrieves the corresponding streams from

the network.

2. Q.931 resides within H.223 and it is a link

layer protocol for establishing connections

and framing data.

H.323 PROTOCOL ARCHITECTURE (cont’d)H.323 PROTOCOL ARCHITECTURE (cont’d)

REGISTRATION, ADMISSION, AND STATUS

1. The H.225 also includes registration, admission, and status

(RAS) control.

2. RAS is the protocol between endpoints and gatekeepers

that makes connections available between them.

CONTROL SIGNALING

1. The H.245 standard provides the call control mechanism

that allows H.323-compatible terminals to connect to each

other.

2. The control messages that it carries relate to: Opening and

closing of logical channels used to carry media streams,

preference requests, flow-control messages and general

commands and indications.

H.323 Protocol Stack

SIPSIPThe Session Initiation Protocol (SIP) is an application layer signaling protocol that defines initiation, modification and termination of interactive multimedia communication sessions between users.It was developed by the IETF and is explained in RFC 2453. It was approved in early 1999.

PhysicalPhysicalPhysicalPhysical

I PI PI PI P

U D PU D PU D PU D P

T C PT C PT C PT C P

S I PS I P

S D PS D P

S I PS I P

S D PS D P

RTP/RTCPRTP/RTCPRTP/RTCPRTP/RTCP

CODECCODECCODECCODEC

D N SD N SD N SD N S

SignalingMediaUtility

HOW DOES SIP MAKE A HOW DOES SIP MAKE A CALL?CALL?

User Registering and Location - determination of the end system to be used for communication

User Availability - determination of the willingness of the called party to engage in communications

User Capabilities - determination of the media and media parameters to be used

Call Setup - ringing and establishing call parameters at both called and calling party

Call Modification – change of media, call forward etc

Call Handling - the transfer and termination of calls

SIP ARCHITECTURESIP ARCHITECTURESIP ARCHITECTURESIP ARCHITECTURE

Intelligent SIP User Agents (UAC/UAS)

Registrar Redirect Location

Proxy Server REGISTER“Here I am”

INVITE“I want to talk to another UA

Proxied INVITE“I’ll handle it for

you”

“Where is this name/phone#?”3xx Redirection

“They moved, try this address”

SIP Gateway

SIP-GW

IP NetworkIP Network

PSTNPSTN

SIP Servers

sip:14083831088@vovida.orgsip:hostname@192.168.10.1

SIP Client

SIP Redirect Server

SIP Client(User Agent Server)

Location Server2

1

RTP Media

45

3

11

7

6

2. bob3. play.com

4. Bob moved. Temporarily contact bob@play.com5. ACK6. INVITE bob@play.com

7. Ringing ok

1. INVITE bob@ieee.org

SIP OPERATION IN REDIRECT MODESIP OPERATION IN REDIRECT MODE

8. ACK

(ieee.org)

(sjsu.edu) (play.com)

8

SIP Client

SIP Redirect & Location Servers

SIP Client(User Agent Server)

SIP Proxy

2. INVITE bob@ieee.org

3

1

RTP Media

SIP Proxy

2

56

4

91211

7

10

3. bob4. play.com

5. Bob moved. Temporarily contact bob@play.com6. ACK7. INVITE bob@play.com 12. ACK

9. Ringing ok

1. INVITE bob@ieee.org

8. INVITE bob@play.com

10. Ringing ok

SIP OPERATION IN PROXY MODESIP OPERATION IN PROXY MODE

8

11. ACK

(Ieee.org)

(sjsu.edu) (play.com)

WHY WAS SIP DESIGNED?WHY WAS SIP DESIGNED?

Flexibility – Does not dictate specifics for architecture, messaging etc. Can even use H.323 URLs to route call.

Scalability and Simplicity – Based on internet model and not single LAN segments. Less storage required.

Ease of creation of new services like buddy lists, instant messaging etc.- Integrating multimedia communications with ease (web-based, email routing mechanisms etc.)

Mobility (Location/Redirect Servers)

Call redirection/forking/multiparty calls ..

COMPARISON OF H.323 and COMPARISON OF H.323 and SIPSIP

VoIP Protocol SIP H.323

Standards Body IETF ITU

Origin Internet/WWW model Telephony model

Complexity/Struct Simple and Modular Complex and Monolithic

Control channel Text based Binary Based

Endpoint Addressing and Call Routing

SIP URL IDRedirect or location servers

H.323 ID AliasAddress mapping mechanism in Gatekeeper

Signaling and Media

UDP and TCP for signaling, RTP for Media

UDP and TCP for signaling, RTP for Media

Conferencing Multicasting. No restrictions on no. of users

Uses MCU for users > 3. Fn. overlaps with RTCP

Client Intelligent User Agents Intelligent H.323 Terminals

Relationship Peer-to-peer Peer-to-peer

Security Registration with Registrar With Gatekeeper

Session Description SDP H.245

MGCPMGCPMedia Gateway Control Protocol is a master-slave protocol that defines communication between telephony Gateways and external call control elements called Media Gateway Controllers or Call Agents.

It was developed by the IETF and explained in RFC 2705. It assumes limited intelligence at endpoints and concentrates it in the core of the network.

Call Agent (master) provides call signaling, control and processing intelligence to the Gatewaysends and receives commands to/from Gateway

Gateway (slave) provides translations between packet and circuit switched networks sends notification to the call agent about endpoint events.

MGCP ARCHITECTUREMGCP ARCHITECTURE

SUPPORTING SUPPORTING PROTOCOLSPROTOCOLS

RSVP RTCP RTP SAP/SDP H.323

SIP

UDP TCP

Underlying Physical, Data Link and Network Layers

SUPPORTING PROTOCOLS SUPPORTING PROTOCOLS (cont’d)(cont’d)

RTP/RTCP (Real-Time Transport & Control Protocols) is used for transporting real time data

RSVP (Resource Reservation Protocol) for reserving resources

RTSP (Real-Time Streaming Protocol) for controlling delivery of real-time media streams

SDP (Session Description Protocol) for advertising multimedia sessions

SAP (Session Announcement Protocol) for describing multimedia session

TECHNICAL ISSUESTECHNICAL ISSUES

Quality of service Delay, jitter, congestion, echo, packet loss, mis-ordered packet arrival

Measure of QoS The mean opinion score is widely used Algorithms: PSQM, PAMS and PESQ

Bandwidth consumption A quality call requires at least 64 kbps. It is impossible to dedicate so much for voice on data network Speech compression techniques are used. For example, silence compression which brings down the bandwidth to 5-6 kbps

TECHNICAL ISSUES (cont’d)TECHNICAL ISSUES (cont’d)

Transparency to the user ease of configuration mapping between IP addresses and phone numbers

Security provides for secure environment using TCP/IP access control can be implemented using authentication calls can be made private using encryption

Security features use four primary components packet filtering router connection gateway address translating firewall application proxy

HARDWARE REQUIREMENTSHARDWARE REQUIREMENTS

Minimum Requirements PC 386 or higher Sound card Full duplex capability Network card or connection to internet or other kind of interface to allow communication between 2 PCs

Companies offering hardware Quicknet, Lucent, 3COM, Cisco, Nortel, Alcatel

Hardware accelerating cards Quicknet PhoneJack Quicknet LineJack VoiceTronix V4PCI VoiceTronix VPB4 VoiceTronix VPB8L

SOFTWARE REQUIREMENTSSOFTWARE REQUIREMENTS

Operating Systems Windows 95, 98, 2000, ME and XP Linux

Gateway Internet Switch Board PSTNGW (Packet Switching Transfer Network Gateway)

Gatekeeper

PRODUCTSPRODUCTS

Gateways: MICOM V/IP Gateway, Nortel Networks CVX SS7 Gateway, Lucent Technologies Pathstar Access Server, Cisco Systems DE-30+ Gateway, 3Com Gateway, VocalTec Series 2000 Gateway, Nuera Solutions Access plus F200 IP

Gatekeepers: Eriksson H.323 gatekeeper, VocalTec Gatekeeper, Nortel Netwroks’ IPConnect, Elemedia H.323 gatekeeper GK2000S

SERVICESSERVICESIP telephones: Cisco's IP phones, Selsius IP phones, Nokia Systems’ IPCourier

PC based software phones: VocalTec IPhone, Netscape’s CoolTalk, Microsoft NetMeeting, WhitePine’s CU-SeeME Pro

FUTURE DEVELOPMENTSFUTURE DEVELOPMENTS

Directory services over telephones

Inter office trunking over the corporate intranet

Remote access to voice, data and fax services of office from home

Fax over IP

Conference bridging

Voice/data synchronization

Text to speech conversion

INTER OFFICE TRUNKING OVER INTER OFFICE TRUNKING OVER CORPORATE INTRANETCORPORATE INTRANET

CONCLUSIONCONCLUSION

VoIP sends voice over data networks instead of data over voice network

Internet along with TCP/IP are driving forces for VoIP technology

Ideal for computer based communications

Market for VoIP is established and is rapidly growing

VoIP cuts communication costs and improves efficiency

Needs QoS for acceptable quality

REFERENCESREFERENCES

www.protocols.comwww.cis.ohio-state.edu/~jain/refs/ref_voip.htm www.iec.org/online/tutorials/vfoip/ www.nwfusion.com/research/voip.html SIP Understanding the Session Initiation Protocol by Alan B. Johnston

Q & A

Thank You!