What is VoIP?• Voice-Over-IP (VoIP)

– Transport of voice traffic using the Internet Protocol (IP)– Not necessarily IP – But only IP provides ubiquitous presence

• VoFR (Voice-over frame relay)• VoATM

– Also referred to as IP Telephony (IPT)• IPT traditionally referred to LAN-based VoIP

• VoIP three major challenges – Voice quality due to available bandwidth – The Internet, inherently, does not provide the best managed network

• No network management• No Quality-of-Service (QoS) solutions• No Service-level agreement (SLAs) between users

– Security issues are also important

A Short Introduction – Transmitting Voice

• The first telephone invented in 1876 by Alexander Graham Bell • The first phone company started in 1879 (AT&T)

– Slogan: A phone for every single household!• Early phones where analog-based with analog transmission

– Noisy when calling long distance – Amplification of the signal caused more noise

• First phone systems where point-to-point– Referred to as POTS (Plain old telephone service)

• Later switches where introduced – PSTN: Public switched telephone network

• Then came digital telephony– Voice signals can be digitized

• The existing telephone system is a circuit-based switched network– Circuits must be established between users prior to their communication – Circuits are in place even if no data is passing through

Analog Technology

Analog TechnologyAnalog Technology

•Trunks: A line or link designed to handle many signals simultaneouslyTrunks: A line or link designed to handle many signals simultaneously•PSTN: Public Switched Telephone Network - Collection of interconnected voice-oriented public telephone networksPSTN: Public Switched Telephone Network - Collection of interconnected voice-oriented public telephone networks•Circuit Switching: Physical path is obtained for a single connection / Connection-orientedCircuit Switching: Physical path is obtained for a single connection / Connection-oriented

SLIC. Subscriber-Loop-Interface-Circuit: A telephone line interface

Call routing in PSTN

• (p.26)

Local Switch Regional Switch Long distance switch regional Local

A Little History – Digital Technology• Analog to Digital Conversion– Pulse Amplitude Modulation (PAM)– Pulse Code Modulation (PCM)– The output of ADC has only two states which are

called Binary in form of 1’s and 0’s• Digital Packets – Small units of data (ones and zeros) routed

through the network with destination address within each packet

Voice Conversion to Digital

A Little History – Packetizing Voice

• The concept of packetized voice goes back to 1974 – No Internet; just sending voice signals in

packet form between two universities• Based on Packet Switching concepts

– Data path is shared and connectionless– No longer dedicated paths as in circuit

switching • First Internet Telephony software

platform (Softphone) introduced in 1994• Initial VOIP systems where phone-centric

– Using the PSTN to get connected to the Internet

Internet

PSTN (Circuit Switching)

Modem Modem

PC PC

Call routing across a VoIP network

• Long distance calls can be carried on the dedicated network• PSTN can be used for local calls• PSTN gateway interface is POTS/T1/DSL/ISDN-PRI

Internet Access using other networks

ISDN –PRI/ interface with the

PSTN along w/Gateway

Renting the dedicated

linesInter

network

Internet

VoIP Motivation

• Network convergence – Having a single infrastructure! – Bringing together two or more diverse networks – Integrating Voice and data networks• Voice is still the killer application / Voice is a major

business! / Largest portion of revenues still come from voice services• Data applications are growing and new services are

emerging

Unifying the network so the Unifying the network so the voice transmission is seamlessvoice transmission is seamless

Why Voice over IP?• Circuit switching was designed for voice

– Expensive yet solid• Today's network

– Many new applications have emerged (email, web, video, IM, etc.)– Higher network flexibility is demanded by providers (mix-and-matching equipments)

• Offering a single network for wide range of applications • IP is an attractive choice for voice transport

– Lower equipment & operation costs• Openness and standardized equipment • Full compatibility • Distributed network rather than centralized

– Integration of voice and data applications • Providing more advanced services • Calling from your web browser?

– Potentially lower BW requirements • VoIP transmission is inherently more transmission efficient • Development of new coding schemes (can also be used for PSTN!)

– Widespread availability of IP • only IP provides ubiquitous presence• Other alternatives are VoFR (Voice-over frame relay) and VoATM

Spend the money on better

circuit-basedTelephone systems?

VoIP Market• According to the second quarter update of the US VoIP Report from TeleGeography.com, 1.23 Million new

customers signed up for wire-line replacement VoIP services during the second quarter of 2006.

www.voipwiki.com/blog/?cat=34

It pays to know about VoIP!

VoIP Challenges

• Offering credible alternative to traditional circuit-switched telephony – High reliability • Five nine availability (99.99999%)

– High quality of voice • Toll-quality • 4.0 or better

– High level of security

ITU-U Recommendation (P.800)1-bad2-Poor3-Fair4-Good 5-Excellent

Speech Quality • Data traffic characteristics– Asynchronized (it can tolerate delay)– Sensitive to packet loss (ACK is required)

• Voice traffic characteristics– Considered as a real time application – Very sensitive to delay – Fewer than 5 percent loss can be tolerated

• Speech quality – Delay – Jitter– Packet loss

Speech Quality - Delay

• Voice packets are very sensitive to delay– Less than 300 msec for telephony– In case of satellite communications:

• 2x120 msec + 200 TCP/IP msec > 300 msec

• Delay is due to packet queuing time – The processor is busy processing other packets – Upper bounds must be established – Shortest path is the path with the least end-to-end

transmission delay time

Delay or Latency Definitions• The time from when words are spoken until

they are heard at the other end– Measure of delay in a call

• Delay is also referred to the time that it takes a packet to make its way through the network to the destination or terminating device

Latency Impact• Large latency values do not necessarily degrade the sound

quality of phone call but large latency values can result in a lack of synchronization between the speakers. This can cause hesitations during the voice conversation make it difficult to interact

• Latency greater than 150 milliseconds is unacceptable in most cases

• One-way latency is used for diagnosing network problems

Latency (Delay)

Factors contribute to Delay• The time it takes for the endpoints to create the packets

used in voice service, known as packet creation latency.• The time it takes to serialize the digital data onto the

physical links of the interconnecting equipment, known as serialization delay.

• The time it takes an electrical signal to travel the length of a conductor, known as propagation delay.

• The time that a network device to buffer a packet and make the forwarding decision, known as packet forwarding delay.

Latency Types

• Fixed delay Codec

Time it takes to sample and digitize the voice signal

Packetization Time it takes to convert voice

into IP packets Network

component propagation due to manufacturing

Jitter buffer

• Variable delay Queuing delay Network delay

Speech Quality - Jitter• Jitter – Defined as delay variation (lack of predictability – high

variance) – way to adjust – Jitter buffers are used to lower the delay variance

• Speech packets are buffered and transmitted at a steady rate – Jitter is due to two factors

• packet routing (Different routs can produce different packet delays)• Different packet queuing time

There is no jitter Problems incircuit-switching!

Example of Jitter

• For example, given a constant packet transmission rate of every 20 ms, new packets would be expected to arrive at the destination exactly over 20ms but unfortunately this is not always the case.

• The figure shows packet one (P1) and packet three (P3) arriving when expected, but packet two (P2) arriving 12ms later then expected and packet four (P4) arriving 5ms late.

Jitter Causes

• The main cause of jitter is queuing variations caused by dynamic changes in network traffic load

• Another cause is equal-cost links do not have the same physical length

Fixing Jitter Impact• The jitter buffer deliberately delays incoming

packets in order to present them to the decompression algorithm at fixed spacing

• The jitter buffer will also fix any out-of-order errors by looking at the sequence number in the RTP frames

Speech Quality – Packet Loss

• Some packets are lost during transmission – Buffer overflow

• Real-time applications cannot utilize the same packet loss avoidance protocols– The communication between the two ends take too long – Retransmission time is very long

• Five percent loss is tolerable

Packet Loss

• VOIP is highly sensitive to packet loss– Loss Rates as low as 1% can garble

communications

• Latency and Jitter can contribute to “virtual packet loss” as packets arriving after their deadline are as good as “lost”

Critical Factors -Overview

• Network management • Speech coding• Network reliability • Network scalability

Network Management

• Network convergence is beneficial but also introduces new challenges– Handling voice and data which have different

characteristics • Network requirements– Voice calls should not be connected if not enough

resources are available • Check sufficient BW

– Support traffic prioritization • Ensure the most critical traffic is least effected when network

congestion occurs • Handle traffic management and QoS

Speech-Coding Techniques

• Choice of speech coding is critical to having high-quality voice

• Two conflicting objectives– Reducing bandwidth – Maintaining the natural-sounding speech (toll

quality)

• A major advantage of VoIP is its distributed characteristic

Reliability & Scalability

• Commercial challenge to to having a telephony network is to ensure five nine availability

• Today’s VoIP systems– Provide sufficient reliability – Enable redundancy and load-sharing

• Good balance between cost and redundancy– Offer scalability

• Scalability refers to supporting higher capacity • Handling millions of simultaneous calls

•

VoIP Standardization Process• Basic architectural issues in VoIP

– Transporting Voice by using IP – Decoding voice

• Various standards have been proposed • Internet standards and specifications are handled by the Internet Society

– Non-profit organization trying to keep the Internet alive and growing – The internet Architecture Board (IAB) – IS advisory group – The Internet Engineering Task Force (IETF) – volunteer who collaborate in the

development of Interne t standards – The Internet Engineering Steering Group (IESG) – responsible for management

of IETF’s activities and their approvals– The Internet Assigned Numbers Authority (IANA) – responsible for the

administration of unique numbers and parameters used in the Internet Standards

Stats

• Today there are over 150 million cell phones in the U.S

• 21 million people registered broadband phones with Skype

Definitions

• Protocols– Set of rules to allow orderly communications

• POTS: Plain old telephone service • PSTN: Public switched telephone network • ITU-T: International Telecommunication Union

Telecommunications Solutions –Standardization Sector

References

• Carrier Grade Voice Over IP, second Edition, D. Collins

Reference – Packet loss and jitter

• www.radcom.com• http://www.voiptroubleshooter.com/

problems/plc.html• http://www.protocols.com/papers/voip2.htm • http://www.juniper.net/solutions/

literature/white_papers/200087.pdf