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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.
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.
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 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 [email protected]. ACK6. INVITE [email protected]
7. Ringing ok
1. INVITE [email protected]
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 [email protected]
3
1
RTP Media
SIP Proxy
2
56
4
91211
7
10
3. bob4. play.com
5. Bob moved. Temporarily contact [email protected]. ACK7. INVITE [email protected] 12. ACK
9. Ringing ok
1. INVITE [email protected]
8. INVITE [email protected]
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.
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
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