August 3-4, 2004 San Jose, CA www.voipdeveloper.com Successfully Offering VoIP- Enabled Applications Services Jan Linden Vice President of Engineering.

Slides:



Advertisements
Similar presentations
CAUSES & CURE OF LATENCY IN THE INTERNET TELEPHONY DR. OLUMIDE SUNDAY ADEWALE Dept of Industrial Math & Computer Science Federal University of Technology.
Advertisements

UBIFone & The Technology Ahead 25 th June 2006 This presentation is the property of UbiFone. Distributors or any other individuals or entities are not.
N Team 15: Final Presentation Peter Nyberg Azadeh Bararsani Adie Tong N N multicodec minisip.
Security in VoIP Networks Juan C Pelaez Florida Atlantic University Security in VoIP Networks Juan C Pelaez Florida Atlantic University.
SIP Simplified August 2010 By Dale Anderson. SIP Simplified Session Initiation Protocol Core of SIP specifications is documented in IETF RFC 3261 Many.
Speech codecs and DCCP with TFRC VoIP mode Magnus Westerlund
© 2006 AudioCodes Ltd. All rights reserved. AudioCodes Confidential Proprietary Signal Processing Technologies in Voice over IP Eli Shoval Audiocodes.
Voice over the Internet (the basics) CS 7270 Networked Applications & Services Lecture-2.
Requirements and Architecture for Zero-Byte Header Compression Pete McCann & Tom Hiller December 13, 2000 draft-mccann-rohc-gehcoarch-00.txt.
© GPRShelp 2004 An Introduction To GPRS. © GPRShelp 2004 Contents What is GPRS? GPRS Applications GPRS Myths GPRS Services – the killer application.
Application Layer 2-1 Chapter 2 Application Layer Computer Networking: A Top Down Approach 6 th edition Jim Kurose, Keith Ross Application Layer – Lecture.
Testing SIP Services Over IP. Agenda  SIP testing – advanced scenarios  SIP testing - Real Life Examples.
Application layer (continued) Week 4 – Lecture 2.
VoIP on the iPhone: Imagine the Possibilities Jan Linden, VP of Engineering.
Successful Multiparty Audio Communication over the Internet Vicky Hardman, M. Angela Sasse and Isidor Kouvelas Department of Computer Science University.
VoIP Voice Transmission Over Data Network. What is VoIP?  A method for Taking analog audio signals Turning audio signals into digital data Digital data.
Nov. 3, 2000 Adaptive Playout Scheduling in Packet Voice Communications.
Introduction to the Application Layer Computer Networks Computer Networks Spring 2012 Spring 2012.
The StarNet Analyzer. Contact SNA Department x172
Successful Multiparty Audio Communication over the Internet Vicky Hardman, M. Angela Sasse and Isidor Kouvelas Department of Computer Science University.
Dr. Philip Cannata 1 Principles of Network Applications.
Leveraging Existing Application Processors in Mobile Devices to Implement VoIP Client.
The Importance of Quality VoIP for Web Conferencing and Collaboration Jan Linden, Vice president of Engineering Global IP Sound, Inc.
VOIP ENGR 475 – Telecommunications Harding University November 16, 2006 Jonathan White.
© Aastra Aastra BluStar for PC High-Quality Audio and HD Video from Your Desktop.
© 2010 Universität Tübingen, WSI-ICS Patrick Schreiner, Christian Hoene Universität Tübingen WSI-ICS 26. July 2010 Rate Adaptation for the IETF IIAC.
N e v e r s t o p t h i n k i n g. IP-Phone solution 2nd Workshop on Wideband Speech Quality in Terminals and Networks: Assessment and Prediction 22nd.
Application Layer 2-1 Chapter 2 Application Layer Computer Networking: A Top Down Approach 6 th edition Jim Kurose, Keith Ross Addison-Wesley March 2012.
VoIP Voice over Internet Protocol
Copyrights © All rights Reserved. Asterisk and VoIP issues Chetan Vaity March 2007.
Voice Over Packet Networks Getting the most from your voice codec Philippe Gournay VoiceAge Corp. 750 Lucerne Road, Suite 250 Montreal (Quebec) H3R 2H6.
SIP Interoperability Testing Alan Percy Director of Business Development AudioCodes, Inc. Booth #822.
Computer Networks: Multimedia Applications Ivan Marsic Rutgers University Chapter 3 – Multimedia & Real-time Applications.
Performance Evaluation of VoIP in Different Settings Tom Christiansen Ioannis Giotis Shobhit Raj Mathur.
1 A high grade secure VoIP using the TEA Encryption Algorithm By Ashraf D. Elbayoumy 2005 International Symposium on Advanced Radio Technologies Boulder,
January 23-26, 2007 Ft. Lauderdale, Florida Host media processing – revisited Faye McClenahan – Aculab.
1 Lab Introduction – software Voice over IP. 2 Lab Capability and Status  Software used in this course installed in Engineering labs including the lab.
DUE Voice over IP (VoIP) Linksys Ernie Friend- FSCJ.
What makes a network good? Ch 2.1: Principles of Network Apps 2: Application Layer1.
Applied Communications Technology Voice Over IP (VOIP) nas1, April 2012 How does VOIP work? Why are we interested? What components does it have? What standards.
August 3-4, 2004 San Jose, CA Migrating from TDM to IP Brough Turner SVP and CTO NMS Communications.
Quality of Service in the Internet The slides of part 1-3 are adapted from the slides of chapter 7 published at the companion website of the book: Computer.
2: Application Layer 1 Chapter 2: Application layer r 2.1 Principles of network applications r 2.2 Web and HTTP r 2.3 FTP r 2.4 Electronic Mail  SMTP,
The Way Forward Factors Driving Video Conferencing Dr. Jan Linden, VP of Engineering Global IP Solutions.
Network Instruments VoIP Analysis. VoIP Basics  What is VoIP?  Packetized voice traffic sent over an IP network  Competes with other traffic on the.
Voice Over Internet Protocol (VoIP). Basic Components of a Telephony Network.
Computer Networks with Internet Technology William Stallings
October 4-7, 2004 Los Angeles, CA VoWLAN Trends and Opportunities Kamal Anand Vice President Marketing Meru Networks
ﺑﺴﻢﺍﷲﺍﻠﺭﺣﻣﻥﺍﻠﺭﺣﻳﻡ. Group Members Nadia Malik01 Malik Fawad03.
Real Time Communications: An Enterprise View Rodger M. Will Ford Motor Company Wednesday, April 21, 2004.
August 3-4, 2004 San Jose, CA DSP vs. HMP Making the right decision August, 2004 Lior Weiss Director Of Product Marketing AudioCodes.
LOG Objectives  Describe some of the VoIP implementation challenges such as Delay/Latency, Jitter, Echo, and Packet Loss  Describe the voice encoding.
Voice over Internet Protocol (VoIP)
Unleashing the Power of IP Communications™ Calling Across The Boundaries Mike Burkett, VP Products September 2002.
Designing Applications Using DSP Modules
Creating Resource-Efficient V2oIP Applications for Low-MHz Mobile Processors Fred Wydler VP VoIP Products SPIRIT DSP.
Quality of Service for Real-Time Network Management Debbie Greenstreet Product Management Director Texas Instruments.
UCA Lync Client for ShoreTel V4.0. Overview Pricing FAQ UCAClient Video Click any UCA logo to return to this page.
Communication Methods
Network customization
VoIP ALLPPT.com _ Free PowerPoint Templates, Diagrams and Charts.
On-Site PBX Vs Hosted PBX.
IP Telephony (VoIP).
VoIP Phones - New era of communication
Managing the performance of multiple radio Multihop ESS Mesh Networks.
Introduction to Networking
ENGR 475 – Telecommunications
WHAT IS MPLS?  MPLS is a type of data-carrying protocol that manages traffic between two locations. It is mainly used in high-performing networks. 
Network customization
Presentation transcript:

August 3-4, 2004 San Jose, CA Successfully Offering VoIP- Enabled Applications Services Jan Linden Vice President of Engineering

August 3-4, 2004 San Jose, CA Message Everything necessary for offering high quality VoIP in the applications market exist today However, many factors have to be considered to avoid pitfalls

August 3-4, 2004 San Jose, CA VoIP in the Applications Space Additions of real-time voice communication in applications are booming –Peer to peer softphones (phone replacement or complement) –Conferencing –Web collaboration –Education (one to many presentations) –Push-to-talk Many different types of devices –PCs –PDAs and smartphones with wireless access –Dedicated hardware

August 3-4, 2004 San Jose, CA Design Considerations Speech quality Ease of use Time-to-market Flexibility Network impairments Cost Signaling Infrastructure Features Device considerations

August 3-4, 2004 San Jose, CA Quality Aspects Attention to details –A chains is only as strong as its weakest link Choice of codec Sampling rate (narrowband 8 kHz vs. wideband 16 kHz) Implementation issues Auxiliary components (e.g. AGC, VAD, AEC) Low latency –One of the most important aspects and also very hard to achieve

August 3-4, 2004 San Jose, CA Handling of Network Degradation Typical problem cases: –Internet applications –Wireless Packet loss concealment –Smooth concealment necessary Jitter –Jitter buffer necessary to ensure continuous playout –Trade-off between delay and quality

August 3-4, 2004 San Jose, CA Ease of use Easy installation and upgrades Audio tuning wizard or similar Hands free operation (requires AEC) Intuitive calling and use of other functionality Presence and address book Support

August 3-4, 2004 San Jose, CA Signaling and Transport Call setup signaling –Proprietary or standard (SIP, H.323, etc) –Reliability –Firewall and NAT traversal –Presence support Transport protocol –Proprietary or standard (UDP, RTP) –Reliability –Firewall and NAT traversal –Real-time performance

August 3-4, 2004 San Jose, CA Auxiliary Functionality PSTN access Voice mail Call forward, hold, etc. Video support Instant Messaging …

August 3-4, 2004 San Jose, CA Infrastructure Server based or peer-to-peer –Call setup –Conferencing –Presence, address book, voice mail, etc. –Compatibility

August 3-4, 2004 San Jose, CA Device Issues Operating system –Is it a real-time OS? –Support for multitasking –Delay and jitter Audio hardware and software –Delay –Quality –Sampling frequency support –Hands free support Clock drift –Hard to detect and compensate for Complexity – especially on PDAs

August 3-4, 2004 San Jose, CA Design Options Build your own –Performance not known - risk –Long development time and time-to-market –Lack of expertise? –High cost –Flexibility Purchase full solution –Known quality –Short time-to-market –Low flexibility –Proven technology

August 3-4, 2004 San Jose, CA Design Options cont. Purchase building blocks –Building blocks: Voice processing Call setup and transport protocols Application –Best of class –Short time-to-market –Flexible –Proven technology

August 3-4, 2004 San Jose, CA Performance Comparison Three VoIP clients with the very similar feature lists compared Significant delay differences –One way delay from 80 ms to 250 ms –Some clients have huge latency variations Basic speech quality –Differences in spectral characteristics (tinny and muffled) –Clicks and pops

August 3-4, 2004 San Jose, CA Performance Comparison cont. Handling of degraded networks –Acceptable packet loss varies from 3 % to 15 % –Long delays Hands-free performance –Many clients have full echo or are cutting out the signal in an unacceptable way Sensitivity to device difference varies Rapidly varying gain control in some cases Complexity fairly similar

August 3-4, 2004 San Jose, CA Performance Comparison cont. Audio demonstration –Same settings for all three clients –Codec used: G.711 –Three test cases: 1.Perfect network 2.3 % packet loss 3.10 % packet loss

August 3-4, 2004 San Jose, CA Performance Comparison cont. Client 1Client 2Client 3 Wideband supportYesNo Echo cancellationYes, full duplex and very good quality for standard setup Yes, very poor performance. Significant clipping and often high echo even for a good setup Yes, very poor performance. Significant clipping high echo 50 % of the time even for a good setup Gain controlYesYes, adapts annoyingly fastYes, starts out with very low volume and adapts annoyingly fast Basic qualityVery good quality, full audio spectrum from ~70 Hz – 4000/8000 Hz Poor, very inconsistent; sometimes very good sometimes a lot of clicks and pops. Spectrum limited at 3500 Hz. Good quality, no audible artifacts but very limited spectral characteristics (300 Hz – 3100 Hz) plus some high frequency noise results in “thin” voice. Packet loss and jitter performance Very robust. 10 % packet loss gives good quality using any codec. Very sensitive to packet loss and jitter both in terms of quality and delay. Quality poor already for 3 percent of packet loss. DelayVery low delay. 80 – 90 ms mouth to ear delay for perfect network. Adds almost no extra delay in jitter buffer when channel is poor. Very high delay. Under perfect conditions one way delay on average 150 ms higher than Client 1 and even more for degraded networks. In worst case more than 500 ms longer delay than Client 1. Very high delay. Under perfect conditions one way delay on average 100 ms higher than Client 1 and even more for degraded networks. In worst case more than 250 ms longer delay than Client 1.

August 3-4, 2004 San Jose, CA Performance Comparison cont. One of the clients consistently outperformed the other two The two worst ones showed similar behavior in most cases –One of the clients had occasional pops and clicks under perfect network conditions This test showed how huge performance differences can be expected depending on the design team’s level of audio expertise

August 3-4, 2004 San Jose, CA Summary High quality VoIP can be offered in the applications market today Lowest cost, shortest time-to-market, and best quality usually possible by purchasing proven VoIP solutions Beware of performance differences between solutions with seemingly identical feature lists