Voip

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Voice Over Internet Protocol (VoIP)

OUTLINE
INTRODUCTION ADVANTAGES OF VoIP POPULAR VoIP PROTOCOLS H.323 SIP MGCP SUPPORTING PROTOCOLS TECHNICAL ISSUES HARWARE REQUIREMENTS SOFTWARE REQUIREMENTS PRODUCTS SERVICES FUTURE DEVELOPMENTS CONCLUSION

INTRODUCTION
VoIP - The ability to carry toll quality voice using compression techniques and packet switching over the IP packet network.

Voice
CODEC: CODEC: Digital to Analog

Voice analog

analog

Analog to Digital

Compress

Decompress

Create Voice Datagram

Re-Sequence and Buffer-Delay

Add Header

digital

(RTP, UDP, IP etc)

Process Header

digital

INTRODUCTION (cont¶d«)
Real time voice traffic can be carried over IP networks in three different ways
1. PC to PC

2.

PC to Phone

3.

Phone to Phone

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 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.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. 2. TERMINALS : Can be either a personal computer or a stand-alone device GATEWAYS : A H.323 gateway provides connectivity between an H.323 network and a nonH.323 network. GATEKEEPERS : Provide call control services such as address translation, bandwidth management, admission control and zone management. 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.

3.

4.

Layout of H.323-enabled inter-network

H.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)
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)
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

SIP
The 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.
Utility Media
CODEC DNS SDP RTP/RTCP UDP IP Physical TCP

Signaling
SIP

HOW DOES SIP MAKE A 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 ARCHITECTURE
SIP Servers

Registrar

Redirect

Location Where is this name/phone#?´

IP Network

3xx Redirection ³They moved, try this address´ REGISTER ³Here I am´ INVITE ³I want to talk to another UA

Proxy Server

sip:[email protected] sip:[email protected]

Proxied INVITE ³I¶ll handle it for you´

SIP Gateway

Intelligent SIP User Agents (UAC/UAS)

SIP-GW
PSTN

SIP OPERATION IN REDIRECT MODE
1. INVITE [email protected] 2. bob 3. play.com 4. Bob moved. Temporarily contact [email protected] 5. ACK 6. INVITE [email protected] 7. Ringing ok 8. ACK

SIP Redirect Server (ieee.org) 2 3 4 5 11 (play.com) Location Server

1

(sjsu.edu) 6 7 8 RTP Media SIP Client

SIP Client (User Agent Server)

SIP OPERATION IN PROXY MODE
1. INVITE [email protected] 2. INVITE [email protected] 3. bob 4. play.com

SIP Redirect & Location Servers (Ieee.org) 3 SIP Proxy 2 5 6 7 (sjsu.edu) 11 10 12 9 8 (play.com) 4 SIP Proxy
9. Ringing ok 10. Ringing ok 11. ACK 12. ACK

5. Bob moved. Temporarily contact [email protected] 6. ACK 7. INVITE [email protected] 8. INVITE [email protected]

1

RTP Media SIP Client SIP Client (User Agent Server)

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 SIP
VoIP Protocol Standards Body Origin Complexity/Struct Control channel Endpoint Addressing and Call Routing SIP IETF Internet/WWW model Simple and Modular Text based SIP URL ID Redirect or location servers H.323 ITU Telephony model Complex and Monolithic Binary Based H.323 ID Alias Address mapping mechanism in Gatekeeper UDP and TCP for signaling, RTP for Media Uses MCU for users > 3. Fn. overlaps with RTCP Intelligent H.323 Terminals Peer-to-peer With Gatekeeper H.245

Signaling and Media UDP and TCP for signaling, RTP for Media Conferencing Client Relationship Security Session Description Multicasting. No restrictions on no. of users Intelligent User Agents Peer-to-peer Registration with Registrar SDP

MGCP
Media 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 Gateway sends 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 ARCHITECTURE

SUPPORTING PROTOCOLS

SIP

RSVP

RTCP

RTP

SAP/SDP

H.323

UDP

TCP

Underlying Physical, Data Link and Network Layers

SUPPORTING PROTOCOLS (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 ISSUES
Quality of service Delay, jitter, congestion, echo, packet loss, misordered 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)
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 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 REQUIREMENTS
Operating Systems Windows 95, 98, 2000, ME and XP Linux Gateway Internet Switch Board PSTNGW (Packet Switching Transfer Network Gateway) Gatekeeper

PRODUCTS
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

SERVICES
IP 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 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 CORPORATE INTRANET

CONCLUSION
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

REFERENCES
www.protocols.com www.cis.ohiostate.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

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