IP TelephonyIP Telephony(Voice Over IP)(Voice Over IP)

Patrick DuffPatrick Duff

Illustrations from Global KnowledgeIllustrations from Global Knowledge

Why Voice Over IPWhy Voice Over IP

Cheaper voice communicationsCheaper voice communications

Offers the ability to combine a voice Offers the ability to combine a voice network with a data networknetwork with a data network

Voice Over IP will offer solutions to Voice Over IP will offer solutions to efficiently send voice (and video) over pre-efficiently send voice (and video) over pre-existing networks existing networks

DefinitionsDefinitions

IP – Internet Protocol; Determines the IP – Internet Protocol; Determines the format of packets and their addresses as format of packets and their addresses as they are sent over the networkthey are sent over the networkRTP – Realtime Transport Protocol; part of RTP – Realtime Transport Protocol; part of voice packet that contains digitized audiovoice packet that contains digitized audioFrame – container of data sent over a Frame – container of data sent over a networknetworkPacket – smaller container sent over a Packet – smaller container sent over a networknetwork

DefinitionsDefinitions

Multiplexers – combine many Multiplexers – combine many communications to be transmitted on one communications to be transmitted on one linelineCodec – encoder/decoderCodec – encoder/decoderPCM – pulse code modulationPCM – pulse code modulationLAN – Local Area NetworkLAN – Local Area NetworkWAN – Wide Area NetworkWAN – Wide Area NetworkPayload – information delivered by the Payload – information delivered by the packet or framepacket or frame

Existing NetworksExisting Networks

Already send out data packetsAlready send out data packets

Maximum Transmission Latency is the Maximum Transmission Latency is the maximum amount of bytes allowed in a maximum amount of bytes allowed in a frame (packet)frame (packet)

Ethernet transmission MTUs are 1500, Ethernet transmission MTUs are 1500, approximately 15 times larger than Voice approximately 15 times larger than Voice MTUsMTUs

So?So?

It takes 214 ms for an Ethernet frame to It takes 214 ms for an Ethernet frame to reach it’s destinationreach it’s destinationA Voice packet has to wait for this frame to A Voice packet has to wait for this frame to finish, exceeding the 150 ms ideal for finish, exceeding the 150 ms ideal for voice communication, and even the 190 voice communication, and even the 190 ms delay for slightly delayed voice ms delay for slightly delayed voice communicationcommunicationThis delay is unacceptable in voice This delay is unacceptable in voice communicationcommunication

SolutionsSolutions

Prioritizing traffic (voice before data)Prioritizing traffic (voice before data)Use Frame Relay Access Device to break Use Frame Relay Access Device to break down data frames into the same size as down data frames into the same size as voice packetsvoice packets The FRAD will then alternate voice and data The FRAD will then alternate voice and data

packets according to prioritypackets according to priority

Upgrading to a T1Upgrading to a T1 Reducing transmission time from 214 ms to Reducing transmission time from 214 ms to

134 ms134 ms

MultiplexersMultiplexers

A voice communications requires only a A voice communications requires only a 4Khz range of frequency4Khz range of frequency

Existing networks can handle a lot more Existing networks can handle a lot more than thisthan this

Multiplexers put multiple communication Multiplexers put multiple communication channels through one wirechannels through one wire

Time Division MultiplexersTime Division Multiplexers

Time Division converts an analog signal Time Division converts an analog signal into a digital signalinto a digital signal

Time Division Multiplexers put each Time Division Multiplexers put each channel into a specific time slotchannel into a specific time slot

Each channel has it’s turn to send it’s Each channel has it’s turn to send it’s signal over the networksignal over the network

Codecs for Voice CompressionCodecs for Voice Compression

““G.711 is a specific PCM scheme that G.711 is a specific PCM scheme that samples voice 8,000 times per second and samples voice 8,000 times per second and converts the 8-bit samples into a digital converts the 8-bit samples into a digital stream.”stream.”Other codecsOther codecs G.729G.729 G.723.1G.723.1

Feeser, Stuart. “Introduction to Voice Over IP”Feeser, Stuart. “Introduction to Voice Over IP”

G.711 PCMG.711 PCM

The G.711 codec uses an 8000 sample The G.711 codec uses an 8000 sample per second clock to convert the 8 bit signal per second clock to convert the 8 bit signal into a digital streaminto a digital stream

At each point of amplitude, the measure is At each point of amplitude, the measure is increased to improve fidelityincreased to improve fidelity

This is called compandingThis is called companding

Companding MethodsCompanding Methods

µ-law - used in North Americaµ-law - used in North America

A-law - used in Europe A-law - used in Europe

µ-lawµ-law

16 logarithmic divisions16 logarithmic divisions

Each contains 16 linearly spaced steps Each contains 16 linearly spaced steps

Only has 255 stepsOnly has 255 steps 0 is not used0 is not used

A-lawA-law

16 logarithmic divisions16 logarithmic divisions

Each contains 16 logarithmically spaced Each contains 16 logarithmically spaced stepssteps

Uses all 256 stepsUses all 256 steps

Size of Voice PacketsSize of Voice Packets

Every byte is not sent out as soon as it is Every byte is not sent out as soon as it is encodedencoded

A packet is created with a 58 byte headerA packet is created with a 58 byte header Two main sizes of packets are usedTwo main sizes of packets are used

20 ms20 ms

60 ms60 ms

20 millisecond packets20 millisecond packets

Collects data for 20 msCollects data for 20 ms

50 packets per second50 packets per second

160 bytes per packet160 bytes per packet

Calculating overheadCalculating overhead 58 bytes * 8 bits per byte * 50 packets per 58 bytes * 8 bits per byte * 50 packets per

second = 23.2 Kbps overheadsecond = 23.2 Kbps overhead

20 millisecond packets20 millisecond packets

BenefitsBenefits Less delay than 60 ms samplingLess delay than 60 ms sampling Good for LANs or WANsGood for LANs or WANs

DisadvantagesDisadvantages More overhead More overhead

50 packets must be sent per second50 packets must be sent per second

60 millisecond packets60 millisecond packets

Collects data for 60 msCollects data for 60 ms

16 2/3 packets per second16 2/3 packets per second

480 bytes per packet480 bytes per packet

Calculating overheadCalculating overhead 58 bytes * 8 bits per byte * 16.7 packets per 58 bytes * 8 bits per byte * 16.7 packets per

second = 7.75 Kbps overheadsecond = 7.75 Kbps overhead

60 millisecond packets60 millisecond packets

BenefitsBenefits Less overheadLess overhead

Only 16 2/3 packets need to be sent per secondOnly 16 2/3 packets need to be sent per second Good for connecting through modemsGood for connecting through modems

DisadvantagesDisadvantages More delayMore delay

58 byte header58 byte header

6 bytes for the MAC address source6 bytes for the MAC address source

6 bytes for the MAC address destination6 bytes for the MAC address destination

4 bytes for IP address originating source4 bytes for IP address originating source

4 bytes for IP address destination4 bytes for IP address destination

2 bytes for Transport Address Source2 bytes for Transport Address Source The port address at the point of originThe port address at the point of origin

2 bytes for Transport Address Destination2 bytes for Transport Address Destination The port address at the destinationThe port address at the destination

58 byte header (cont.)58 byte header (cont.)

1 byte for priority of the packet1 byte for priority of the packet

2 bytes for the sequence number2 bytes for the sequence number Where this packet belongs in the total number Where this packet belongs in the total number

of packetsof packets

1 byte for the HOP counter1 byte for the HOP counter How many routers this packet went throughHow many routers this packet went through

2 bytes for the Payload type2 bytes for the Payload type What this packet is carryingWhat this packet is carrying

58 byte header (cont.)58 byte header (cont.)

4 bytes for the Timestamp header4 bytes for the Timestamp header4 bytes for the synchronization source4 bytes for the synchronization source IP address of the computer that created this IP address of the computer that created this

packetpacket

1 byte for the Version1 byte for the Version Version number of the itemsVersion number of the items

3 bytes for the Protocol ID3 bytes for the Protocol ID Rules governing hardware address, network Rules governing hardware address, network

address and transport addressaddress and transport address

58 byte header (cont.)58 byte header (cont.)

4 bytes for flags4 bytes for flags

4 bytes for validity of the payload 4 bytes for validity of the payload (checksum)(checksum)

4 bytes for length4 bytes for length

4 bytes for frame check sequence4 bytes for frame check sequence Payload validity checked at each router or Payload validity checked at each router or

ethernet switchethernet switch

Packet SwitchingPacket Switching

Voice Packets containVoice Packets contain

RTP = Real Time ProtocolRTP = Real Time Protocol

UDP = User Datagram ProtocolUDP = User Datagram Protocol

IP = Internet ProtocolIP = Internet Protocol

MAC = Media Access ControlMAC = Media Access Control

When a packet is sent outWhen a packet is sent out

Each part of the Each part of the header directs the header directs the packet to it’s packet to it’s destinationdestination

Voice Over IP in ActionVoice Over IP in Action

Convert voice to dataConvert voice to data

Send out data, start creating new packetSend out data, start creating new packet

What happens when Net Traffic is heavy What happens when Net Traffic is heavy and packets arrive out of order or in and packets arrive out of order or in different intervals than sent out?different intervals than sent out?

The Jitter BufferThe Jitter Buffer

Collects the packetsCollects the packets

Each packet is time stamped according to Each packet is time stamped according to the 8000 tick per second clock as to when the 8000 tick per second clock as to when the packet was createdthe packet was created

The jitter buffer will collect all the packets The jitter buffer will collect all the packets and detect time irregularities between and detect time irregularities between packets with a synchronized 8000 tick per packets with a synchronized 8000 tick per second clock second clock

The Jitter BufferThe Jitter Buffer

So packets arrived too fast, too slow, or So packets arrived too fast, too slow, or out of orderout of order

Step 1, reorder packetsStep 1, reorder packets

Step 2, determine slowest arriving packetStep 2, determine slowest arriving packet

Step 3, determine delay of packetStep 3, determine delay of packet

Step 4, delay all other packets the same Step 4, delay all other packets the same amount of timeamount of time

Silence SuppressionSilence Suppression

Most of the time during voice Most of the time during voice communications, one member is not communications, one member is not talkingtalking

Instead of sending empty packets, don’t Instead of sending empty packets, don’t send any packetssend any packets

After not sending packets, how do we re-After not sending packets, how do we re-synchronize with the jitter buffer?synchronize with the jitter buffer?

Silence SuppressionSilence Suppression

20 ms packet example:20 ms packet example:Each packet should arrive every 160 ticksEach packet should arrive every 160 ticksThe packets aren’t out of order, but have The packets aren’t out of order, but have skipped some timeskipped some timeThe jitter buffer looks at the time stamp The jitter buffer looks at the time stamp and delays the packet according to the and delays the packet according to the amount of time between the last packet’s amount of time between the last packet’s timestamp and the current packet’s timestamp and the current packet’s timestamptimestamp

RTP MixersRTP Mixers

RTP mixers combine multiple incoming RTP mixers combine multiple incoming signals, then sends them to all intended signals, then sends them to all intended receiversreceivers

Source timing isn’t always synchronizedSource timing isn’t always synchronized

The RTP mixer decides it’s own time The RTP mixer decides it’s own time stamping and sends the combined signal stamping and sends the combined signal with a new time stampwith a new time stamp

Why RTP Mixers?Why RTP Mixers?

A conference call with 4 callersA conference call with 4 callers

One voice signal goes to the other 3One voice signal goes to the other 3

If two people talk at the same timeIf two people talk at the same time The mixer mixes the two voices and sends it The mixer mixes the two voices and sends it

to the two silent peopleto the two silent people The two people talking don’t get the mixed The two people talking don’t get the mixed

signal, only the signal of the other person signal, only the signal of the other person talkingtalking

TranslatorTranslator

If two people on the same Voice Over IP If two people on the same Voice Over IP call use two different size packetscall use two different size packets 60 ms60 ms 20 ms20 ms

The translator on each side will accept the The translator on each side will accept the incoming data even though it’s sending out incoming data even though it’s sending out data in a different formatdata in a different format

TCPTCP

Transfer Control ProtocolTransfer Control Protocol

3 step connection setup3 step connection setup

Sends 1 packet, waits for Sends 1 packet, waits for acknowledgementacknowledgement

Sends 2 packets, waits for Sends 2 packets, waits for acknowledgementacknowledgement

Number of packets increases Number of packets increases exponentiallyexponentially

TCP (cont.)TCP (cont.)

Number of packets increases Number of packets increases exponentially until a threshold is reachedexponentially until a threshold is reached

At this threshold, the number of packets At this threshold, the number of packets sent reverts back to one and then increase sent reverts back to one and then increase linearly until a new lower threshold is linearly until a new lower threshold is determineddetermined

This second threshold is the connection This second threshold is the connection speedspeed

TCPTCPGood for data, bad for voiceGood for data, bad for voice

Too much delay in the 3 step setupToo much delay in the 3 step setup

The acknowledgement confirms that data The acknowledgement confirms that data is transferred successfully, but creates too is transferred successfully, but creates too much delay for voice transfermuch delay for voice transfer

UDPUDP

User Datagram ProtocolUser Datagram Protocol

No connection acknowledgementNo connection acknowledgement

No transfer acknowledgementNo transfer acknowledgement

Send packets and forgets themSend packets and forgets them

Since 5% packet loss is acceptable for a Since 5% packet loss is acceptable for a voice communication, UDP is idealvoice communication, UDP is ideal

Basics of TelephonyBasics of Telephony

TransmissionTransmission Transfer of data from one end to anotherTransfer of data from one end to another

SwitchingSwitching Converting data from one medium to anotherConverting data from one medium to another Analog to DigitalAnalog to Digital

SignalingSignaling Setting up and tearing down a callSetting up and tearing down a call

Making a CallMaking a Call

Dial a numberDial a number

Sends a message to your Private Branch Sends a message to your Private Branch eXchangeeXchange

Sends a message to your Central OfficeSends a message to your Central Office Generates Initial Address MessageGenerates Initial Address Message

Connects to CO of receiving endConnects to CO of receiving end Sometimes through Tandem COSometimes through Tandem CO

Sends message to PBX of receiving endSends message to PBX of receiving end

Rings the phone on the receiving endRings the phone on the receiving end

Making a CallingMaking a Calling

SignalingSignaling

SS7 (System Signaling 7)SS7 (System Signaling 7) Allows a telephone network to setup route and Allows a telephone network to setup route and

control callscontrol calls 3 major components of SS73 major components of SS7

SCP (Signal Control Point)SCP (Signal Control Point) Helps determine how to route a callHelps determine how to route a call

SSP (Signal Switching Point)SSP (Signal Switching Point) Starts, teminates and combines callsStarts, teminates and combines calls

STP (Signal Transfer Point)STP (Signal Transfer Point) Routes network traffic between switchesRoutes network traffic between switches

SignalingSignaling

AddressingAddressing

E.164 AddressingE.164 Addressing NANP North American NANP North American

Numbering PlanNumbering Plan Area codeArea code CO numberCO number Line NumberLine Number

ConclusionConclusion

FREE LONG DISTANCE CALLING!FREE LONG DISTANCE CALLING!

Existing networks are barely able to carry Existing networks are barely able to carry VOIP communicationsVOIP communications

More developments compress the voice More developments compress the voice communication more and more, as communication more and more, as networks keep improvingnetworks keep improving

Possibly the new way to make phone Possibly the new way to make phone calls?calls?