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Speech Coders – a VoIP perspective. Roar Hagen CTO SIP/email: roar.hagen@globalipsound.com. 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
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Speech Coders – a VoIP perspective Roar HagenCTO SIP/email: roar.hagen@globalipsound.com
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
percentage of respondents 0 10 20 30 40 50 60 source: Forrester Research/AT&T (2000) QoS – statusIndustry’s perspective quality concerns unproven technology PSTN works fine too busy to switch not compelling economics
PSTN Managed network Wireless Managed network Public Internet [ ] Next generation codecs should address the needs of all applications Background - Diverse Environment
Wireless VoIP – The Big Unknown ? Mobility Vehicle 2G/2.5G 3G WLAN Walk Fixed LAN Bluetooth 0.1 10 1 Mbps 100
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
Current speech processing technology not designed for packet switched environments “FALL OFF A CLIFF” shape of curve forces over provisioning MOS = USER EXPERIENCE OVERPROVISIONED NETWORK CONGESTED NETWORK * MEAN OPINION SCORE
…congestion related VoIP QoS problems can be solved without over provisioning… MOS = USER EXPERIENCE OperateAT and ABOVE congestion point without customer knowing OVERPROVISIONED NETWORK CONGESTED NETWORK * MEAN OPINION SCORE
GIPS Ehanced G.711+ GIPS NetEQ™ G.711+GIPS NetEQ™ G.711+ITU PLC G.729A G.711+No PLC 0% 5% 10% 15% 20% 25% 30% 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 4.5 4.0 4.0 3.5 3.5 MOS MOS 3.0 3.0 2.5 2.5 GIPS iPCM™-wb+ 2.0 2.0 GIPS NetEQ™-wb G.722+ GIPS NetEQ™-wb 1.5 1.5 G.722.1 Source + no PLC 1.0 1.0 0% 5% 10% 15% 20% 25% NETWORK CONDITION (% PACKET LOSS) NETWORK CONDITION (% PACKET LOSS) SOURCE LOCKHEED MARTIN GLOBAL TELECOMMUNICATION (COMSAT)
Jitter Buffer/PLC Enhancements Source: Lockheed Martin Global Telecommunications (COMSAT)
140 Jitter Adaptive jitter buffer Fixed jitter buffer NetEQ™ 120 100 80 Delay (ms) 60 40 20 0 0 200 400 600 800 1000 1200 1400 1600 1800 2000 Packet number Delay gain with NetEQ™approx. 30-60ms compared to traditional jitter buffers
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 Source: Dynastat Inc.
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 - codec spec (exper. stds track) • draft-duric-rtp-ilbc-00.txt - RTP payload profile (AVT group) • Web site www.globalipsound.com • Free demo SIP client available, please request at: SIP/email: alan.duric@globalipsound.com
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