Voice over IP (VoIP)

by

Kiran Kumar Devaram

Varsha Mahadevan

Shashidhar Rampally

What’s VoIP?

• VoIP is the ability to make telephone calls and send faxes over IP-based data networks with a suitable quality of service and superior cost/benefit.

Motivations for VoIP

• Demand for Multimedia communication

• Demand for integration of Voice and Data networks

• Cost Reduction in long distance telephone calls

How to VoIP?

Analog D ig ita l Vo ice

Compression to less than 32Kbps

Transfers through Routers, LAN Switches etc, using their Protocols

Voice To/From IPAnalog

Digital

Voice

CODEC: Analog to Digital

Compress

Create Voice Datagram

Add Header(RTP, UDP, IP, etc)

N etw ork

Voice To/From IP

Digital

Analog

Process Header

Re-sequence and Buffer Delay

Decompress

CODEC: Digital to Analog

N etw ork

Voice

Configuration OptionsTelephone-to-Telephone

PC-to-PC

Telephone-to-PC

ISO Reference Model and VoIP Standards

 ISO Protocol layer  Protocols and standards

Presentation  Codecs / Applications

Session  H.323 / SIP / MGCP

Transport  RTP / TCP / UDP

Network  IP

Link  FR, ATM, Ethernet, PPP, etc.

 

VoIP Standards

• ITU– H.323

• IETF– Session Initiation Protocol (SIP)– Media Gateway Control Protocol (MGCP)

H.323 Entities

LAN

Terminal

Terminal

Gateway

Gatekeeper

MCU

Terminal

• Endpoint on a LAN• Supports real-time, 2-way communications with another

H.323 entity• Must support:

– Voice - audio codecs– Signaling and setup

• Optional support:– Video– Data

Gateway

• Interface between the LAN and the circuit switched network

• Translates communication procedures and formats between networks

• Call setup and clearing• Compression and packetization

of voice• Example: IP/PSTN gateway

Gatekeeper

• The most vital component of H.323 system• Manages a zone (a collection of H.323 devices)• Usually one gatekeeper per zone; alternate gatekeeper might

exist for backup and load balancing

• Functionalities:– Address Translation– Call authorization and signaling– Bandwidth Management– Call Management

Multi-point Control Unit (MCU)

• Endpoint that supports conferences between 3 or more endpoints

• Can be stand-alone device (e.g., PC) or integrated into a gateway, gatekeeper or terminal

• Typically consists of

-multi-point controller(MC)

-multi-point processor(MP)

H.323 Protocol StackTransfer of real-time media (audio and video)

Registration

Control and Signaling

H.323 Call Stages

• Discovery and Registration(RAS) – Who am I• Call Setup(RAS/H.225/Q.931) – Whom I want to call• Call Negotiation (H.245) – These are our capabilities• Media Channel Setup(H.245) – Let’s open audio channel• Media Transport( RTP/RTCP) – Send audio datagrams• Call termination (H.245/H.225/RAS) – We are done

Simple VoIP CallCaller Number : 785-537-2736

Called Number : 410-944-511

ITSP Number : 1-888-745-2654

Local Loop Trunk

785-537-2736

Local Switch

Gateway

1-888-745-2654

Caller dials ITSP toll free number : 1-888-745-2654

Caller gets connected to VoIP gateway of ITSP

Simple VoIP Call

785-537-2736

Local Switch

Gateway

1-888-745-2654

What is the IP address of the destination gateway for 410-944-2511?-LRQ

The IP address of the destination gateway is 154.23.78.345. – LCF

May I call the IP address? ARQ

You may use XX Kbps bandwidth - ACF

Gatekeeper

ARQ

ACFLRQLCF

Simple VoIP Call

785-537-2736

Local Switch

Gateway

1-888-745-2654

The setup message consists of

Originator gateway IP address (129.130.10.123) Destination Gateway IP address (154.23.78.345)

Caller-number (785-537-2736) Called-number (410-944-2511)

H.245 request: OpenLogicalChannelForAudio

Gatekeeper

Connect H.225/Q.931/H.245

Destination Gateway

Simple VoIP Call

785-537-2736

Local Switch

Gateway

1-888-745-2654

Destination gateway makes a request to the gatekeeper to accept the call from the originator

May I call the originator gateway IP address? ARQ

Yes,You may use XX Kbps bandwidth - ACF

Gatekeeper

ARQ

ACF

Destination Gateway

Simple VoIP Call

785-537-2736

Local Switch

Gateway

1-888-745-2654

Destination gateway sends a connect confirm message.

Gatekeeper

Connect H.225/Q.931/H.245

Destination Gateway

Simple VoIP Call

Local Switch

Gateway

Gatekeeper

Local SwitchGateway

Destination Gateway establishes PSTN connection with PSTN circuit switch and H.245 audio channel

Caller will hear the ringer tone generated by the destination switch

SIP: Session Initiation Protocol

• It’s a signaling protocol proposed by IETF.• Establish sessions.• SIP is a text-based, peer-to-peer protocol that runs on the Session Layer.• SIP Address Format (resembles mailto: URL format)

– sip:henrys@wcom.com

– sip: +1-972-555-1234@wcom.com; user=phone

• Integrated heavily w/ Internet technologies such as web (http), email & messaging services, and directory services (LDAP, DNS).

• Location Independent and hence opted for Mobile Networks.

SIP Architecture

• Major Entities– User Agent– Intermediate Server

• Proxy Server

• Redirect Server

– SIP Registrar

SIP Architecture (contd.)

• User Agents– User Agent Client (UAC)– User Agent Server (UAS)

• Registrar ( resembles a DNS )A Registrar matches the SIP address with the IP address.

SIP Proxy Operation

SIP Client

CallerSIP Client

Callee

SIP Proxy Server

1. SIP Clients registers with SIP servers at login or at boot up

2. When user picks up phone and dials destination phone number or URL, request is sent to the proxy server

3. Proxy server looks up phone number or URL to registered called party, SIP server then sends invitation to called party

4. Called Client is informed of incoming call by an invitation from proxy server

5. SIP Clients open RTP session between themselves when the called user picks up the phone

SIP Redirect Operation

SIP Client

CallerSIP Client

Callee

SIP Redirect server

1. SIP Clients registers with SIP servers at login or at boot up

2. When user picks up phone and dials destination phone number or URL, request is sent to the redirect server

3. Redirect server looks up phone number or URL to registered called party, SIP server then sends the address back to the call originator

4. Call originator sends invitation to destination

5. Called client is informed of incoming call by invitation message (Phone ring)

6.SIP Clients open RTP session between themselves when the called user picks up the phone

H.323 vs SIP

H.323 SIP

Philosophy Designed for multimedia communication over different types of networks

Designed to open a session b/w two points

Reliability Designed to handle failure of network entities

No defined procedures for handling device failure

Message Encoding Encodes in compact binary format

Encodes in ASCII text format. Hence easy to debug and process

Addressing Flexible addressing scheme using URLs and E.164 numbers

Understands only URLs style addresses

Architecture Monolithic Modular

QoS Issues

Delay For high quality voice, one way latency must not be greater than 150ms. Delay greater than 50ms leads to echo and talker overlap.

Jitter Variation in inter-packet arrival time. The solution to this problem is to introduce jitter buffers.

Packet Loss Loss in excess of 5-10% causes significant degradation in voice quality.

Re-ordering Packets may arrive out of order and this leads to garbled speech.

Voice enabled Software

• NetMeeting, WindowsMessenger (Microsoft)• Net2Phone CommCenter (Net2Phone)• DialpadChameleon (DialPad)• eDial Desktop Voice Conferencing System (eDial) • IP Communications (WorldCom)

Future of VoIP

• In the year 2000, VoIP networks carried 1 percent or $700 million of total voice traffic.

• This level will grow to 13 percent by 2003, and have a value at that time of $24 billion.

• The established carriers in the U.S. generated some $83 billion carrying long-distance traffic in the year 2000.

• This figure will drop by $6 billion to approximately $77 billion in 2002.• Many believe that the whole idea of per-minute rates will disappear and,

within two years, flat rates will prevail for long distance just as they do for Internet access, thanks to VoIP!!!!!

Case Study

Migrating the CIS department network to a multi-service network

Multi-Service Networking

It is the integration of data, voice, and video networks.

&

VoIP is a subset of the same.

Phases of Multi-Service Migration

• Readying the network infrastructure for real-time traffic

• IP Telephony, or Desktop Telephony, involves installing IP Phones, voice-capable computer applications and Web-based multimedia applications that integrate voice and data to the desktop

Specifications

• 1 PBX

• 6 POTS lines

• 50 Extensions– 40 extensions are connected to the staff– 10 extensions are connected to student labs

Existing Network Topology

Router CNS Router

CNS Router

N etw ork

CIS LAN CNS LAN

PSTN Volume and ExpensesType # of

PeopleAvg. Mins per day per person

% of Internal Calls

% of International calls

Work

days per month

Total Mins per month per type

Cost per min for USA call

Cost per min for International call

Monthly cost per type

Staff 40 120 10% 0.1% 21.67 104000 $0.07 $0.54 $6600

Lab 10*20 5 0.5% 0% 30 30000 - - $500

Total $7100

Voice Traffic Calc.

• 2 hours call volume per staff user per day X 40 users + 1/12 hours call volume per lab user per day X 200 users = 97 hours daily call volume

• 97 hours X 60 minutes per hour = 5820 minutes per day • 5820 minutes X 17% (busy hour load) = 990 minutes per busy

hour • 990 minutes per busy hour X 1 Erlang/60 minutes per busy hour =

16.5 Erlangs • 16.5 Erlangs X 90% of out-bound traffic = 14.85 Erlangs volume

proposed• Number of trunks reqd. for 14.85 Erlangs = 28

Bandwidth Considerations

• 6 out going lines require a maximum of 144Kbps

• CIS dept. has a bandwidth >100Mbps !

CISCO Multi-Service Equipment

• Cisco 2610 modular access router

• 28 key system FXO trunks connected to it

Financial AnalysisEquipment Estimated Cost (in US dollars)

Cisco 2610 Modular Access Routers

$2,918

PBX Trunk Module $5,418

Key System Modules $7,700

Total Capital Cost $16,036

Financial Analysis

Monthly PSTN Voice Savings $7,100

Net Total Annual Savings $85,200

Capital Costs $16,036

Installation (Estimate) $3,000

Total Capital Costs $19,036

Payback Period (Months) 2.68 !!!

Conclusion

VoIP is the way to go !

References

• Voice over IP (ISBN: 0-13-022463-4) – Uyless Black

• http://www.protocols.com/papers/voip.htm

• http://www.networkmagazine.com/encyclopedia/search?term=IPtelephony

• ftp://ftp.netlab.ohio-state.edu/pub/jain/courses/cis788-99/voip_protocols/index.html

• http://members.tripod.com/taegon/voip/current_problems.htm

• http://www.itpapers.com/techguide/voiceip.pdf • http://www.zdnet.com/products/stories/reviews/

0,4161,2626792,00.html

Questions ?