What is VoIP ?

Voice Service using Internet Protocol over best effort broadband internet, Public IP carrier or Private IP Network

Lower Cost for end users

GreaterEfficiency

Higher Reliability

Supporting Innovation

Concept Voice over PSTN Voice over IP

Switching Circuit switched Packet switched

Bandwidth Dedicated Dynamically allocated

Latency <100ms 200-700ms

Quality of service High Low and variable

Comparision of voice over PSTN and voice over IP

Ringing

Talk

Quality of Service (QoS) is about bandwidth (and latency) management.

Packet loss

Network delay

Delay jitter

Throughput

Network availability

QoS

RFC 3714

IAB concerns affected by persistent, high packet drop rates that would arise from rapid growth in best effort telephony on best effort networks

Congestion Collapse

User Quality

Fairness

Example of potential for trouble

Georgia,Atlanta

Nairobi, Kenya

Access Links

128 Kbps last hop bottleneck

TCP data traffic(1500B packet size) + VoIP traffic produces

additional 90 ms delay for some packets which is above ITU-T

recommended time for speech traffic.

Congestion Collapse

Occur in networks with flows that traverse multiple congested links having persistent, high packet drop rates.

All traffic slows to a crawl and nobody gets acceptable packet delivery or acceptable performance.

Avoidance

Use end to end congestion control

Call rejection (not available for best-effort traffic)

User Quality

2

1

N

1

2

N

64 Kbps64 Kbps

128 Kbps

Arrival Rate = N * 64 KbpsSuccessful transmission = 2 * 64 KbpsDrop rate = (N-2) /N

Drops occur randomly, and none of the flows can be expected to present better quality service to users.

Solution: End-to-end congestion control be used by each VoIP, and

use a codec that can adapt the bit rate to the bandwidth actually received by that flow.

Rule of thumb

When packet loss rate > 20 %

Audio quality of VoIP is degraded beyond usefulness due to bursty nature of traffic.

Fairness

Considering TCP flows sharing connection with VoIP flows, VoIP crowds out TCP.

Possible solution:

Allocate bandwidth on congested links to classes of traffic.

VoIP traffic should not be exempt from end to end congestion control.

Congestion control for real time traffic

Current effort in IETF

RTP (Upgrade)

TFRC

DCCP

Adaptive rate Audio Codecs

RTP (RFC 3551)

Supports the transport of real-time media, including voice traffic, over packet networks which suppresses silence conditions .Contain media information and a header, providing information to the receiver that allows the reordering of any out-of-order packets.Uses payload identification to describe the encoding of the media so that it can be changed in light of varying network conditions.Encoding Scheme for audio

Sample based encoding (DVI4, G722)Frame based encoding (G723, GSM)

RTP

In order to carry RTP in protocols offering a byte stream abstraction, such as TCP the application MUST define its own method of delineating RTP and RTCP packets.

RTP data SHOULD be carried on an even UDP port number and the corresponding RTCP packets SHOULD be carried on the next higher (odd) port number.

No specification of security services. Possibility of DOS attack for data encodings using compression techniques that have non-uniform receiver-end computational load.

TFRC

Lower variation of throughput over time compared to TCP, thus suitable for applications such as telephony or streaming media.

Designed for applications that use a fixed packet size, and vary their sending rate in packets per second in response to congestion.

TFRC-PS is a variant of TFRC for applications having a fixed sending rate but vary their packet size in response to congestion.

DCCP

Specified for unreliable flows, with the application being able to specify either TCP-like or TFRC congestion control.

Congestion control identifiers

CCID 2 for TCP-like congestion control

CCID 3 for TFRC congestion control.

CCID 4 for TFRC-PS congestion control (future use).

Adaptive Rate Audio Codecs

Operates at a low sending rate, or reduces the sending rate as throughput decreases and/or packet loss increases.

Improves the scalability of VoIP or TCP sharing a congested link

Effective use of available bandwidth.

Example: AMR-WBsupports eight speech encoding modes having bit rates between 4.75 and 12.2 kbps.

Adaptive Rate Audio Codecs

Reduces transmission rate during silence periods.

Supports audio from different channels to be separately encoded and decoded each of the individual channels.

Unequal Bit-error Detection and Protection.

Employs forward error correction (FEC) and frame interleaving, to increase robustness against packet loss.

VoIP protocols

H.323

MGCP (Media Gateway control Protocol)

Megaco H.248

SIP

SAPv2

H.323 Architecture

System Control Unit

Provides call control and framing capabilities.

Includes following standardsH.225.0

Q.931

H.245

Audio and Video Codecs

Define the format of audio and video information and represent the way audio and video are compressed and transmitted over

the network .Required audio and video codecs

G.711G.723H.261H.263

G.723 and H.263 are default codecs preferred for NetMeeting connections, which offer the low-bit rate connections necessary for audio and video transmission over the Internet.

Components of H.323

H.323 Terminal

Gatekeeper

MCU

Gateway

H.323 Terminal

ISDN

PSTN

H.332,H.321,H.310 Terminals

Additional Units to H.323 Terminal Architecture

TerminalsAre LAN client endpoints that provide real-time, two-way communications.

Multipoint Control Unit (MCU)Allows three or more H.323 terminals to connect and participate in a multipoint conference

GatewaysMakes H.323 terminals on a LAN available to H.323 terminals on a wide area network (WAN) or another H.323 gateway.

GatekeepersProvides network services to H.323 terminals, MCUs, and gateways

H.323 Protocol Stack

References

RFC 3714 RFC 3551 RFC 3267 Network convergence and voice over IP

(Technology Review#2001-2) Understanding the H.323 Standard