Transcript Speech Coders – a VoIP perspective Roar Hagen CTO SIP/email: [email protected] Agenda • Speech Coders – a VoIP perspective • Demo • Q&A.

Speech Coders – a VoIP perspective
Roar Hagen
CTO
SIP/email: [email protected]
Agenda
• Speech Coders – a VoIP perspective
• Demo
• Q&A
QoS – (endpoints) status
”A lot of talk, ... but not much work”
• Year after year the same story
• More then 3000 papers since 1984
• Limited ToS support at the end points
QoS – status
Industry’s perspective
percentage of respondents
0
10
20
30
40
50
60
quality concerns
unproven technology
PSTN works fine
too busy to switch
not compelling economics
source: Forrester Research/AT&T (2000)
Background - Diverse Environment
PSTN
Managed
network
[
Managed
network
Public
Internet
]
Next generation codecs should
address the needs of all applications
Wireless
Packet Loss San Francisco – Hong Kong
Jitter San Francisco – Hong Kong
Homg Kong to China VoIP Call
2G/2.5G
Wireless VoIP – The Big Unknown ?
3G
Fixed
Walk
Vehicle
Mobility
WLAN
LAN
Bluetooth
0.1
1
10
100
Mbps
Approach
We need holistic view/approach for both
• Horizontal (end-to-end) perspective
• Vertical (top-down) perspective
Vertical (Top Down) Perspective
Presentation
Speech Codecs/…
Session
SIP/H.323
Transport
RTP/UDP/RSVP
Network
IP/WFQ/IP-prec
Link
MLPPP/FR/ATM AAL1
Physical
VoIP Aspirations
• IP innovation rather than PSTN replication
• New features and services through voice and data
convergence
• End-to-end IP
• Better than PSTN sound quality
MOS = USER EXPERIENCE
Current speech processing technology not
designed for packet switched environments
“FALL OFF A CLIFF”
shape of curve forces
over provisioning
OVERPROVISIONED
NETWORK
CONGESTED NETWORK
*
MEAN OPINION SCORE
MOS = USER EXPERIENCE
…congestion related VoIP QoS problems can
be solved without over provisioning…
Operate AT and ABOVE
congestion point without
customer knowing
OVERPROVISIONED
NETWORK
CONGESTED NETWORK
*
MEAN OPINION SCORE
narrow band
sound quality equal to
PSTN
wide band
sound quality
Better Than PSTN Quality
Matching PSTN Quality
Telephony bandwidth speech test result
Wideband speech
5
5
4.5
4.0
4.0
3.5
3.5
MOS
MOS
4.5
3.0
2.5
GIPS Ehanced G.711+
GIPS NetEQ™
2.0
G.711+GIPS NetEQ™
G.711+ITU PLC
1.5
3.0
2.5
GIPS iPCM™-wb+
GIPS NetEQ™-wb
2.0
G.722+
GIPS NetEQ™-wb
G.722.1
1.5
G.729A
1.0
0%
G.711+No PLC
5%
10%
15%
20%
25%
30%
NETWORK CONDITION (% PACKET LOSS)
1.0
0%
Source + no PLC
5%
10%
15%
20%
25%
NETWORK CONDITION (% PACKET LOSS)
SOURCE LOCKHEED MARTIN GLOBAL TELECOMMUNICATION (COMSAT)
Jitter Buffer/PLC Enhancements
Source: Lockheed Martin Global Telecommunications (COMSAT)
Delay gain with NetEQ™
approx. 30-60ms compared to traditional jitter buffers
Jitter
140
Adaptive jitter buffer
NetEQ™
Fixed jitter buffer
Delay (ms)
120
100
80
60
40
20
0
0
200
400
600
800
1000
1200
Packet number
1400
1600
1800
2000
The NextGen Speech Codec Ideal
•
Need one concept that will work for a long time
– footprint importance
•
Need to handle large diversity of transport network
–
–
–
–
–
•
•
low rate
high quality, high rate
packet loss
jitter
low delay
Manageable IPR situation
Signal Robustness
– speech
– music
•
Suitable for variety of applications, e.g. IP video-conferencing
iLBC (internet Low Bitrate Codec)
• Speech sampled at 8 kHZ,
• using a block-independent linear-predictive coding (LPC)
algorithm.
• Bandwidth 13.867 kbps (52 bytes per 30 ms)
• Frame size 30 ms (support for 20 ms in the next revision)
• Complexity and memory requirements are similar to ITU
G.729A
• Basic Quality is equal to or better than G.729. Packet loss
robustness is significantly better than G.729.
• Packet loss concealment - Integrated example solution
MOS Results
G.729A
G.723.1
iLBC
4.0
MOS
3.5
3.0
2.5
2.0
1.5
0
Source: Dynastat Inc.
5
10
Packet Loss [%]
15
iLBC - IETF work
• IETF deliverables, submitted during February ‘02:
– iLBC codec specification draft
- experimental
standards track
– iLBC RTP Payload Profile
- regular standards
track (AVT)
– Statement about IPRs in iLBC and its “freeware nature”
• MOS results submission to the AVT mailing list during
March ‘02
Why iLBC !?
•
Current low bit rate codecs: ITU G.729, G.723.1, GSM-EFR, and
3GPP-AMR were developed for circuit switched & wireless
telephony and are all based on the CELP (Code Excited Linear
Prediction) paradigm.
•
CELP coders are stateful, they have memory, error propagation
results from lost or delayed packets.
•
iLBC treats every packet individually, making it suitable for
packet communications.
More information
• Coming Soon - web site www.ilbcfreeware.org with:
–
–
–
–
–
Info about initiative
Info about codec
Latest iLBC IETF drafts (spec and payload format)
Latest iLBC float point Source code
FAQ list
• IETF drafts:
– draft-andersen-ilbc-00.txt
– draft-duric-rtp-ilbc-00.txt
- codec spec (exper. stds track)
- RTP payload profile (AVT group)
• Web site www.globalipsound.com
• Free demo SIP client available, please request at:
SIP/email: [email protected]
Summary
• Current speech coding technology not suited for VoIP
• VoIP opens possibilities
– Move quality exprience to the next level with wideband coders
• NGN will not be NGN unless we move step forward on all of its
fields
• iLBC – internet Low Bit Rate Codec
– Provide an open standard ”the Internet way” for coder
Demo