Introduction to Voice over IP (VoIP)

Sean Riley

Partner Marketing Manager

Grandstream Networks, Inc.

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Introduction

• VoIP Acronyms• Basics of VoIP• Comparisons with current phone systems• VoIP Protocols• VoIP Issues• Telephone Regulation• Future of VoIP• Deployment Scenario

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VoIP Acronyms

ATA Analog Telephone adapter  A device that coverts analog voice signals to digital signals which can then be transmitted over the Internet. 

Auto Attendant  An automatic response system, such as a voice presenting options such as press 2 for sales, 5 for Lisa, etc., which handles incoming calls and sends them to the appropriate phone or message.

BYOD Bring Your Own Device  Some VoIP providers allow a person to supply their own equipment or ATA.

CDR Call Detail Recording  Details about a specific call that includes duration, origination, destination, and billable information, as well as other pertinent information.

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VoIP Acronyms (Page 2)

Codec Normally used to reference to converting analog signals to digital or digital signals to analog.  It can be used in conjunction with compression software to compress and decompress these signals to varying degrees.

DHCP Dynamic Host Control Protocol  A communications protocol that lets network administrators supervise and distribute IP addresses from a central point to each computer or device on a network.

DID  Direct Inward Dialing  A service that allows an enterprise to allocate individual phone numbers to each person within its PBX system.

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VoIP Acronyms (Page 3)

DTMF  Dual Tone Multi-frequency Also known as Touchtone, it is the signal generated when you press a telephone's touch keys that is sent to the telephone company.  These signals are actually two tones of a specific frequency designed so that a voice cannot duplicate them.  The ability for interactive telephone menus to work correctly across different networks and phone systems is due to the fact that DTMF tones are standardized and are uniquely linked to a number (and # or *) on the telephone keypad.

Echo Cancellation  Echo cancellation is the process of eliminating echo from voice communication to improve the quality of the call. It is necessary because speech compression techniques and packet processing delays generate echo, of which there are 2 types, acoustic echo and hybrid echo. Echo cancellation improves voice quality in VoIP calls and also reduces the required bandwidth due to silence suppression techniques.

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VoIP Acronyms (Page 4)

IVR  Interactive Voice Response  An integrated software information system that speaks to callers and uses menus and voice responses.  By using touch-tone keypad entries to interact with the software, you get voice responses with real time data.

Jitter   As data load increases and decreases, routers on the Internet can create slightly different times that individual packets take to travel from one point to another point.  This variation in time is known as jitter.

Latency   The time it takes for a packet to reach its destination.  Higher delay times can be an issue, especially for VoIP, where voice delay can be recognized with latency higher than 150 milliseconds.  Higher than 500 milliseconds and the conversation is going to be very problematic.

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VoIP Acronyms (Page 5)

PBX  Public Branch Exchange  A private telephone switching system that allows outside phone lines from a telecommunications provider to connect to extensions within the office or building.  They usually have multiple features including call forwarding, rollover, paging and voice mail.

POTS Plain Old Telephone System  The familiar single phone line, single phone number system that has been in existence for many years.

PSTN Public Switched Telephone Network  The combination of local, long-distance and international carriers that make up the worldwide telephone network.

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VoIP Acronyms (Page 6)

SIP Session Initiation Protocol  is a signaling protocol for Internet conferencing, telephony, and instant messaging.  It is a request-response protocol, dealing with requests from clients and responses from servers. initiating an interactive user session.

SIP Trunking is the use of VoIP to facilitate the connection of typically a PBX to the Internet, where the Internet replaces the conventional telephone trunk, allowing a business to communicate with traditional PSTN telephone subscribers worldwide by connecting to an ITSP (Internet Telephony Service Provider). 

Softphone   IP telephony software that allows end users to send and receive calls over the computer or hand held PC device (PDA) over the Internet.  Typically used in conjunction with a headset and microphone

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VoIP Acronyms (Page 7)

STUN  Simple transversal of UDP through NATs  is a protocol for assisting devices behind a NAT firewall or router with their packet routing.

Switch  A switch is a device that keeps a record of the MAC addresses of all devices connected to it and then channels incoming data from any of multiple input ports to the specific output port that will take the data toward its intended destination.

T38 A recognized codec standard for sending fax transmissions over an IP network in real time. Messages are sent as UDP or TCP/IP packets.

UDP User Datagram Protocol is a communications protocol that does not provide sequencing of the packets. The application must be able to make sure that the entire message has arrived and is in the right order.

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VoIP Basic Technology

• Voice over Internet Protocol (VoIP) is the process of digitally sending an audio signal (usually voice) over a network or the internet

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How Phones Used To Work

• Phones were connected by a direct physical line from 1 user to another user.

• The line remained open for the entire duration of the call.

• A call from California to New York required a large length of copper wire connecting the 2 talkers. It would cost a lot for the call because the talkers would actually own a 3,000 mile long copper wire for the period of time they were on the phone.

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How Analog Works Today

• Your voice is digitized and sent along with other users across a fiber optic cable.

• These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. So in a 10-minute conversation, the total transmission is 9,600 KB.

• If you look at a typical phone transmission much of the transmitted data is wasted. While you are talking the other party is listening (and vice versa) such that only half the line is used at any time.

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How IP Phones Work

• IP Phones receive phone numbers then determine if the number is an IP “soft phone” or a physical “hard phone” by contacting the VoIP company’s Call processor.

• If it a soft phone then an IP address is acquired, otherwise the call will go through the VoIP company’s phone network.

• IP phones data is sent as packets and only when there is data to be sent i.e. talking.

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Comparison Example

• An important difference between the IP phones and an analog phone services is the amount of redundant data sent.

Jack

Jill

JackJill

Hi Hi

How are You?

How are You?

Great!!! Great!!!

Analog Phone Implementation IP Phones

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Network Protocol Structure for Internet Telephony

H.323 or SIPRTP, RTCP, RSVP, RTSP

Transport Layer (UDP, TCP)Network Layer (IP, IP Multicast)

Data Link LayerPhysical Layer

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Session Initialization Protocol

• Application layer In charge of establishing and terminating sessions in Internet telephony

• Client's Methods

– Invite - invites callee(s) to participate in a call

– ACK - Acknowledges the invitation

– Options - Inquires about media capabilities without setting up a call

– Cancel - Terminates the invitation

– Bye - Terminates a Call

– Register - Sends users Location to a registrar (a SIP server)

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Functionality of IP Phones

They come with all the regular phone options including:

– Caller ID– Call waiting – Call transfer– Repeat dial– Return call – Three-way calling – Call Forwarding (with additional features)

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Current Implementationsof IP Telephony

• IP Phone

– Soft (software

– Hard (hardware; physical device)

• Analog Telephone Adapter (ATA)

• Analog Gateway

• Private Branch Exchange (PBX)

• Online Video Gaming

• Online Messaging (Skype, Google Talk)

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IP Phone

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ATA

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Analog Gateway

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IP PBX

Things to understand(as an IP-based technology)

1) Privacy

2) Ensure correct billing

3) Protection from Denial of Service (DoS)

4) Protection from packet manipulation.

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Ways to Secure VoIP

1) Encrypt VoIP traffic

2) Properly configure firewalls

3) Consider segmenting voice and data traffic by using a virtual LAN

4) Use proxy servers in front of corporate firewalls to process incoming and outgoing voice data

5) Make sure that server based IP PBXs are locked down and protected against viruses and DDoS attacks.

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VoIP – Minimum requirements

• Broadband internet connection• Willingness to venture into IP, (while saving 50-

80% vs. analog phones)

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Why VoIP ???

1) Typically much easier & cheaper to install

2) Save Money (often more than 50%!)

3) Easier to upgrade

4) Keep your voice and its pertinent data on the same network

5) Different devices can communicate with each other

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VoIP vs. PSTN

• Carrier Lines

• Bandwidth

• Added Features

• Expandability and Upgradeability

• Typical Cost of a Business Line Rental

• Typical Local Call Cost

• Free Calls

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VoIP Advantages

• Reliability (Redundancy)

– QoS

– PoE

• Security

• Phone Regulations/Cost

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Quality of Service

• Improvement for any time-dependent internet application

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Power over Ethernet

• Useful application in the deployment of VoIP phones• Power over Ethernet (PoE) allows devices to be

powered by the ethernet connection• Advantages

– Allows use in remote areas– Can connect to a network backed up by a UPS

so when power outages occur, phone is still powered and functional

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VoIP and Telephone Regulations

• VoIP telephone service is a direct competitor with the traditional phone service.

• This new use for the Internet is deemed a “internet service” rather than a “voice service”.

• This distinction allows for cost savings due to passing ‘country tolls’

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VoIP Regulation

• $$$ is not the only force behind transitioning to VoIP

• Regulation has given way to standards of service including, but not limited to 911 availability

• As of May 19, 2005, the FCC has required that all VoIP providers deliver all 911 calls to the customer’s local emergency operator.

– Customers must be able to update their local information at any time, even remotely.

– Applicable to any phone that connect to the PSTN

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College Deployment

Analog telephone wiring being routed

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Analog Phone Connections

Wiring to the outside

Telephone connections to Patch Panel

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Deployment Scenario:Seeing the light…

• Currently deploys more than 40 IP phones to remote sites

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How do I get VoIP ??!!

• Consumer Providers– Vonage– Basic Talk– Ring Central– Vitelity

• Business Providers– Phone Power– Broadvox– Bandwidth.com

Questions??