Voice over IP: Issues

and Challenges

IP

Raj Jain
The Ohio State University
Columbus, OH 43210
[email protected]
http://www.cis.ohio-state.edu/~jain/
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 Overview

q Voice over IP: Why?

q Sample Products and Services
q 13 Technical Issues
q 4 Other Issues
q Protocols
q H.323 Standard
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 Market
q International VOIP calls could cost 1/5th of normal
rates ⇒ Big share of $18B US to foreign calls.
$15B within Europe.
q 500,000 IP telephony users at the end of 1995.
q 15% of all voice calls on IP/Internet by 2000
⇒ 10M users and $500M in VOIP product sales in
1999 [IDC]
q US VOIP service will grow from $30M in 1998 to
$2B in 2004 [Forester Research]
$2B in 2001 and $16B by 2004 [Frost & Sullivan]
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 Scenario 1: PC to PC
IP
Network

q Need a PC with sound card

q IP Telephony software: Cuseeme, Internet Phone, ...
q Video optional

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 Scenario 2: PC to Phone

IP Phone
Network Network
Gateway

q Need a gateway that connects IP network to phone

network (Router to PBX)

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 Scenario 3: Phone to Phone
Phone IP Phone
Network Network Network
Gateway Gateway

q Need more gateways that connect IP network to phone

networks
q The IP network could be dedicated intra-net or the
Internet.
q The phone networks could be intra-company PBXs or
the carrier switches
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 Advantages
A C A D

B D B C
q Private voice networks require n(n-1) access links.
Private data networks require only n access links.
q Voice has per-minute distance sensitive charge
Data has flat time-insensitive distance-insensitve
charge
q Easy alternate routing ⇒ More reliability
q No 64kbps bandwidth limitation
⇒ Easy to provide high-fidelity voice
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 Applications
q Any voice communication where PC is already used:
m Document conferencing

m Helpdesk access

m On-line order placement

q International callbacks
(many operators use voice over frame relay)
q Intranet telephony
q Internet fax

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 Sample Products
q VocalTec Internet Phone: PC to PC.
q Microsoft NetMeeting: PC to PC. Free.
q Internet PhoneJACK: ISA card to connect a standard
phone to PC. Works with NetMeeting, InternetPhone
etc. Provides compression.
q Internet LineJACK: Single-line gateway.
q Micom V/IP Family:
m Analog and digital voice interface cards

m PC and/or gateway

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 Products (Cont)
PBX Gateway

PC w Router IP Network
V/IP S/w
m Features:
q Compression

q Phone number to IP address translation.

q Supports RSVP.

q Limits number of calls.

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 Products (Cont)
q VocalTec Internet Telephony Gateway:
m Similar to Micom V/IP

m Interactive voice response system for problem

reporting
m Allows WWW plug in

m Can monitor other gateways and use alternate

routes including PSTN
m Sold to Telecom Finland. New Zealand Telecom.

q Lucent's Internet Telephony Server: Gateway|

Lucent PathStar Access Server
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 Products (Cont)
q CISCO 2600 Routers: Voice interface cards (VICs)
Reduces one hop.
q Baynetworks, 3COM, and other router vendors have
announced product plans

PBX Router IP Network

PC
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 Sample Services
q IDT Corporation offers Net2Phone, Carrier2Phone,
Phone2Phone services.
q Global Exchange Carrier offers international calls
using VocalTec InternetPhone s/w and gateways
q Quest offers 7.5¢/min VOIP Q.talk service in 16
cities.
q ITXC provides infrastructure and management to
'Internet Telephone Service Providers (ITSPs)'
q America On-line offers 9¢/min service.
q AT&T announced 7.5¢/min VOIP trials in 9 US cities.
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 Services (Cont)
q Other trials: USA Global link, Delta 3, WorldCom,
MCI, U.S. West, Bell Atlantic, Sprint, AT&T/Japan,
KDD/Japan, Dacom/Korea, Deutsche Telekom in
Germany, France Telecom, Telecom Finland, and
New Zealand Telecom.
q Level 3 is building a nation wide IP network for
telephony.
q Bell Canada has formed 'Emergis' division.
q Bellcore has formed 'Soliant Internet Systems' unit
q Bell Labs has formed 'Elemedia' division
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 Technical Issues
1. Large Delay
m Normal Phone: 10 ms/kmile ⇒ 30 ms coast-to-
coast
m G.729: 10 ms to serialize the frame + 5 ms look
ahead + 10 ms computation = 25 ms one way
algorithmic delay
m G.723.1 = 100 ms one-way algorithmic delay

m Jitter buffer = 40-60 ms

m Poor implementations ⇒ 400 ms in the PC

m In a survey, 77% users found delay unacceptable.

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 Technical Issues (Cont)
2. Delay Jitter: Need priority for voice packets.
Shorter packets? IP precedence (TOS) field.
3. Frame length: 9 kB at 64 kbps = 1.125 s
Smaller MTU ⇒ Fragment large packets
4. Lost Packets: Replace lost packets by silence,
extrapolate previous waveform
5. Echo cancellation: 2-wire to 4-wire.
Some FR and IP systems include echo suppressors.
IP/Phone
PBX Network PBX

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Reflection Reflection Raj Jain
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 Technical Issues (Cont)
6. Silence suppression
7. Address translation: Phone # to IP. Directory servers.
8. Telephony signaling: Different PBXs may use different
signaling methods.
9. Bandwidth Reservations: Need RSVP.
10. Multiplexing: Subchannel multiplexing
⇒ Multiple voice calls in one packet.
11. Security: Firewalls may not allow incoming IP traffic
12. Insecurity of internet
13. Voice compression: Load reduction
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 Other Issues
1. Per-minute distance-sensitive charge vs
flat time-insensitive distance-insensitive charge
2. Video requires a bulk of bits but costs little.
Voice is expensive. On IP, bits are bits.
3. National regulations and government monopolies
⇒ Many countries forbid voice over IP
In Hungary, Portugal, etc., it is illegal to access a web
site with VOIP s/w. In USA, Association of
Telecommunications Carriers (ACTA) petitioned FCC
to levy universal access charges in ISPs
4. Modem traffic can’t get more than 2400 bps.
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 Compression Standards
q G.711: 64 kbps Pulse Code Modulation (PCM)
q G.721:
m 32 kbps Adaptive Differential PCM (ADPCM).

m Difference between actual and predicted sample.

m Used on international circuits

q G.728: 16 kbps Code Excited Linear Prediction

(CELP).
q G.729: 8 kbps Conjugate-Structure Algebraic Code
Excited Linear Prediction (CS-ACELP).

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 Compression (Cont)
q G.729A:
m A reduced complexity version in Annex A of
G.729.
m Supported by AT&T, Lucent, NTT.

m Used in simultaneous voice and data (SVD)

modems.
m Used in Voice over Frame Relay (VFRADs).

m 4 kbps with proprietary silence suppression.

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 Compression (Cont)
q G.723.1: Dual rates (5.3 and 6.3 kbps).
m Packet loss tolerant.

m Silence suppression option.

m Recommended by International Multimedia

Teleconferencing Consortium (IMTC)'s VOIP
forum as default for H.323.
m Supported by Microsoft, Intel.

m Mean opinion score (MOS) of 3.8.

4.0 = Toll quality.

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 Protocols
q RSVP: Resource Reservation protocol [RFC 2205]
q RTP: Real-time Transport Protocol.
Sequencing, timestamp, payload identification, and
delivery monitoring. [RFC 1889]
q RTCP: RTP Control Protocol. Provides delivery
feedback.
q RTSP: Real-time Streaming Protocol. Allows
controlling streaming audio/video. [RFC 2326]
q SIP: Session Initiation Protocol [IETF mmusic]
q SDP: Session Description Protocol [RFC 2327]
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 Protocols (Cont)
q SAP: Session Announcement Protocol [IETF mmusic]
q SCCP: Simple Conference Control Protocol [IETF
mmusic]
q VPIM: Voice Profile for Internet Mail [RFC 1911].
Voice messages in email. MIME profile.
q SCTP: Simple Computer Telephony Protocol. Like
SMTP for mail. [www.phonezone.com]
q 802.1p: Priority over LANs
q 802.1Q: Virtual LANs

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 Protocols (Cont)
q S.100: Standard application programming interface
for computer telephony.
m Endorsed by the Enterprise Computer Telephony
Forum.
m Will allow applications from different vendors to
share the telephony server resources for switching,
routing, and media processing.
q SCbus: High-speed TDM bus for computer telephony.
m Endorsed by ANSI.

m Telephony products from Micom, VocalTec, IDT

are all based on SCbus.
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 Protocols (Cont)
q Internet Fax Routing standard: Allows routing
communication among fax servers.
q H.323 Internet telephony (video conferencing)
standard

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 Protocols (Cont)
q IPv6: 4-bit priority, 24-bit flow label
q IP over ATM: MPOA allows QoS.
q MPLS: Multiprotcol Label Switching. Will support
QoS. [IETF mpls]
q ST-II: Stream Protocol V2. Connection oriented IP.
IPv5. Provides resource reservations. [RFC 1819]
q Integrated Services: Guaranteed (CBR) and
controlled-load (nrt-VBR) services. [RFC 2211+2212]
q Multicasting: IGMP [RFC 2236]
q Multicast Routing: MOSPF, DVMRP, PIM
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 Telephony/Conferencing Systems

Multiplexing/Demultiplexing
Video I/O Video
Equipment Codec

Audio I/O Audio

Equipment Codec
Network
Interface Network
Data Data
Application Protocol
System Control
Control Protocol
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 Conferencing Standards
Network ISDN ATM PSTN LAN POTs
Conf. Std. H.320 H.321 H.322 H.323 V1/V2 H.324
Year 1990 1995 1995 1996/1998 1996
Audio G.711, G.711, G.711, G.711, G.723.1,
Codec G.722, G.722, G.722, G.722, G.729
G.728 G.728 G.728 G.723.1,
G.728, G.729
Audio Rates 64, 48-64 64, 48-64, 64, 48-64, 64, 48-64, 16, 8, 5.3/6.3
kbps 16 16 8, 5.3/6.3
Video H.261 H.261, H.261, H.261 H.261
Codec H.263 H.263 H.263 H.263
Data Sharing T.120 T.120 T.120 T.120 T.120
Control H.230, H.242 H.242, H.245 H.245
H.242 H.230
Multiplexing H.221 H.221 H.221 H.225.0 H.223
Signaling Q.931 Q.931 Q.931 Q.931 -

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 H.323 Protocols
q Multimedia over LANs
q Provides component descriptions, signaling
procedures, call control, system control, audio/video
codecs, data protocols
Video Audio Control and Management Data
G.711, G.722, H.225.0 H.225.0 H.245 T.124
H.261 RTCP
G.723.1, G.728, RAS Signaling Control
H.263
G.729
RTP X.224 Class 0 T.125
UDP TCP T.123
Network (IP)
Datalink (IEEE 802.3)
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 H.323 Components
Terminals Gatekeeper MCU

ISDN Firewall H.323

Router Proxy
PSTN

Router To Internet
ATM

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Gateway Raj Jain
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 H.323 Terminals
Video I/O H.261
Equipment H.263
G.711 RTP
Audio I/O

LAN Interface
G.723
Equipment G.729
Data
T.120
Application
H.245 Control
System
Control Q.931 Call Setup
User RAS Gatekeeper
Interface Interface
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 H.323 Terminals
q Client end points. PCs.
q H.245 to negotiate channel usage and capabilities.
q Q.931 for call signaling and call setup.
q Registration/Admission/Status (RAS) protocol to
communicate with gatekeepers.
q RTP/RTCP for sequencing audio and video packets.

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 H.323 Gateways
q Provide translation between H.323 and other terminal
types (PSTN, ISDN, H.324)
q Not required for communication with H.323 terminals
on the same LAN.

Gateway
H.323 Protocol ISDN
Terminal Translation Terminal
Processing and Interworking Processing

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 H.323 Gatekeepers
q Provide call control services to registered end points.
q One gatekeeper can serve multiple LANs
q Address translation (LAN-IP)
q Admission Control: Authorization
q Bandwidth management
(Limit number of calls on the LAN)
q Zone Management: Serve all registered users within
its zone of control
q Forward unanswered calls
q May optionally handle Q.931 call control
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 H.323 MCUs
q Multipoint Control Units
q Support multipoint conferences
q Multipoint controller (MC) determines common
capabilities.
q Multipoint processor (MP) mixes, switches, processes
media streams.
q MP is optional. Terminals multicast if no MP.
MCU
Multicast Unicast

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 SIP
q Session Initiation Protocol
q Allows locating a callee at different locations
q Callee registers different locations with SIP Server
q Servers can also use finger, rwhois, ldap to find a callee
q Messages: Ack, Bye, Invite, Register, Redirection, ...
Location
X Server Jain@cis Jain@acm
Invite Jain@cis
Moved to Jain@acm
Invite Jain@acm
Ack Jain@acm
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 Summary

q Voice over IP products and services are being rolled out

q Ideal for computer-based communications
q IP needs QoS for acceptable quality
q A number of working group at IETF are working on it
q H.323 provides interoperability
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 Organizations
q IMTC: International Multimedia Teleconferencing
Consortium
q VOIP: Founded in 1966 by Cisco, Microsoft,
VocalTec, 3Com/USR, Dialogic etc. to augment
H.323. Folded into IMTC.
q Enterprise Computer Telephony Forum (ECTF),
http://www.ectf.org
q Internet Fax Routing Forum
q Voice Profile for Internet Mail (VPIM) Work Group
of EMA
q VON Coalition, Inc., http://www.von.org
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 IETF Working Groups
Multimedia:
q IP Telephony (iptel): RFC 1789 on INETPhone
servers. Will develop Gateway attribute distribution
protocol and call processing syntax.
q Internet Fax (fax): Data representation, addressing,
and transport of faxes over IP.
q PSTN and Internet Interworking (pint):
m Initiation of telephone services from IP hosts.

m Web users can request call back, fax, fax-back

services.
m Phone users can request web pages (via speech).
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 IETF (Cont)
q Audio/Video Transport (avt): Real-time transmission
of audio and video over UDP and IP multicast. RTP,
RTCP.
q Multiparty Multimedia Session Control (mmusic):
Internet Teleconferencing. SDP, SAP, RTSP, SIP,
SCCP.
q Also several working groups on multicasting and QoS

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 References
q See
http://www.cis.ohio-state.edu/~jain/refs/ref_voip.htm
for a detailed list of references.

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 Thank You!

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