Voice and Desktop Video Teleconferencing over IP

Willis Kim

9 July 2005

Application Voice QualityOnly Level Possible

in a PBX

Wideband

G.72264-256k

G.723.16.3k

G.729a8k

PSTN/PBX

G.71164k

Digital Signal Processor Advances

Bigger,Cheaper Bandwidth

IP Precedence, 802.1p,IntServ/RSVP, DiffServ

VoiceData

Voice

IP Telephony (and Video) Enablers

Data Exceeds VoiceOver Voice Net

Network ConsolidationUsage Costs

Trunk Charges

DSP DSP

QoS Advances

Data

H.323/MGCP/SGCP/SIP

FRF.11/12

AAL1/2/5

Open TelephonyStandards

Reason for Convergence on IP Convergence Delivers

Reduced maintenance costs via common infrastructure

Increased productivity & Decreased Time-to-Deploy

Better leverage of capital across multiple organizations

Greater administration and operation efficiencies

Single Infrastructure

Packet-BasedMultiservice Network

DataDataVoiceVoice

VideoVideo

InternetInternet

Multiservice Networks

Integration of data, voice, and video services into a single packet-based infrastructure using IP

Both in enterprise and public service provider networks infrastructure

Open Standards H.323 - An International Telecommunications Union (ITU)

interoperability protocol enabling cross-communication of multimedia products and applications over packet-based networks. Under H.323, multimedia products offered by one vendor can work with those of another, regardless of hardware compatibility. For example, a PC can share audio and video streams over either an intranet or the Internet. Applications are thus network-, platform-, and application-independent. Extension of ITU-T standard H.320 that enables videoconferencing over LANs and other packet-switched networks, as well as video over the Internet.

SIP - Session Initiation Protocol (SIP) is a peer-to-peer, multimedia signaling protocol developed in the IETF. SIP is ASCII-based, resembling HTTP, and reuses existing IP protocols (DNS, SDP, etc.) to provide media setup and teardown. Since its first publication in 1999, SIP has generated a high level of interest in the VoIP industry, and many people believe that SIP will become the de facto standard protocol for future voice networks.

Non-Open Standards

SCCP (Skinny Call Control Protocol) - Skinny gateways are a series of digital gateways that include the DT-24+, the DT-30+, and the WS-X6608-x1 Catalyst voice module.

Connectivity

Call ProcessingCluster

PSTNPSTN

PSTNPSTN

WANWAN

Call ProcessingCluster

Switch

PSTNPSTN

Switch Switch

PSTN – Public Switched Telephone NetworkPSTN – Public Switched Telephone Network

How can I check my Internet connection?

VoiceVoice Data

Data

Bandwidth Low toModerate

Low toModerate

Moderateto High

Moderateto High

Random Drop Sensitive LowLow HighHigh

Delay Sensitive HighHigh LowLow

Jitter Sensitive HighHigh LowLow

Typical Application QoS Requirements

Firewall and NAT Refresher

Ethe

rnet

Inside Firewall

Ethe

rnet

Client

ClientServer

Connection requests

Connection replies

Connection Table Connection ConnectionConnection Connection ConnectionConnection Connection Connection

Network Address Translation (Overloading)

Why do firewalls or NAT cause problems?

With H.323, SIP and SCCP heavily use dynamically allocated ports

Egress is ok, but igress is not

Why is QoS Important? In an Agency Network QoS:

Ensures that real-time traffic gets priority Ensures that ERP/business-critical traffic is

serviced appropriately Ensures that non-critical traffic

(Napster, Gnotella, etc.) does NOT consume valuable bandwidth

Protects against denial of service attacks!

Packet Labeling - Layer 3

4 4 8 16 16 3 13 8 8 16 32 32 bits Hdr Total Header Source DestinationVers Len TOS Length ID Flags Offset TTL Prot Chksum Address Address Data...

Precedence D T R Unused

3 1 1 1 2

IP V4 Packet – IP Precedence

bits

4 4 8 16 16 3 13 8 8 16 32 32 bits Hdr Total Header Source DestinationVers Len DS Length ID Flags Offset TTL Prot Chksum Address Address Data...

DSCP Unused

6 2

IP V4 Packet - IETF DiffServ

bits

3 bits = 8 values; 0-73 bits = 8 values; 0-73 bits = 8 values; 0-73 bits = 8 values; 0-7

6 bits = 64 values; 0-636 bits = 64 values; 0-63

VoIP and Video Demonstration Cisco IP Phone

Configuration Unity Call Manager Express on Cisco 3825

(gateway) IP Phone – 7960 and 7920

Call and Analysis Video

End-points TANGBERG 1500 D-Link DVC-1000