VOIPmageddon. Are We Headed for a Telephony Meltdown? Phil Edholm President and Principal PKE Consulting LLC

Transcription

1 VOIPmageddon Are We Headed for a Telephony Meltdown? Phil Edholm President and Principal PKE Consulting LLC

2 About The Presenter Phil Edholm is the President of PKE Consulting LLC. PKE consults to end users and vendors in the communications and networking markets. Phil is the organizer of WebRTC World Conference & Expo, the global eco-system event for WebRTC. Phil Edholm has over 30 years experience in creating innovation and transformation in networking and communications. Prior to founding PKE, Phil was Vice President of Technology Strategy and Innovation for Avaya and CTO/CSO for the Nortel Enterprise business. Phil is recognized as an industry leader and visionary. In 2007, he was recognized by Frost and Sullivan with a Lifetime Achievement Award for Growth, Innovation and Leadership in Telecommunications for creating the VoIP industry. Phil is a widely sought speaker and has been in the VoiceCon/Enterprise Connect Great Debate three times. He has been recognized by the IEEE as the originator of Edholm s Law of Bandwidth as published in July 2004 IEEE Spectrum magazine and as one of the Top 100 Voices of IP Communications" by Internet Telephony magazine. Phil was a member of the IEEE standards committee, developed the first multi-protocol network interfaces, and was a founder of the Frame Relay Forum. Phil has 16 patents and holds a BSME/EE from Kettering University. UC Summit - PKE Consulting

3 Is this the new telephony reality? UC Summit - PKE Consulting

4 Topics Why is VoIP quality degraded? Why is this an issue now? What can we do to avoid VoIPmageddon? UC Summit - PKE Consulting

5 VoIP can be very good or very bad Peer to peer VoIP can be excellent Packets directly from one peer to the other CoS/QoS adds packet delivery assurance Inbound echo can manage echo well HD voice can be added However, most VoIP calls today do not follow this model UC Summit - PKE Consulting

6 Causes of VoIP Quality Issues Latency Multi-IP Reverse hauling Both echo and Human CSMA/CD/Interrupt Transcoding Echo Remember that TDM was Less than 15 msec round trip local loop latency no echo cancellation Less than 200 msecs round trip latency anywhere in the world no Human CSMA/CD/Interrupt Constant time domain easy echo cancellation G711 end to end UC Summit - PKE Consulting

7 VoIP Endpoint Scenarios off premise dialing TDM to TDM Old Paradigm VoIP to TDM 200X Paradigm VoIP to VoIP 201X Paradigm Direct VoIP End to End IP Only ing VoIP (TDM or SIP) UC Summit - PKE Consulting

8 Human Echo Human CSMA/CD Human Patience VoIP is Driven by Our Perceptions of Latency Human Time Public Switched Telephony Network Data Network Open Internet Human Tolerance of Delay Network Time Echo becomes perceptible at about msecs Human CSMA/CD/Interrupt is perceptible at about msecs Patience can be trained Time in Seconds UC Summit - PKE Consulting

9 Human Echo Human CSMA/CD Human Patience VoIP is Driven by Our Perceptions of Latency Human Time Public Packet Telephony Network Data Network Open Internet Human Tolerance of Delay Echo becomes perceptible at about msecs Human CSMA/CD/Interrupt is perceptible at about msecs Patience can be trained Time in Seconds UC Summit - PKE Consulting

10 Percentage of Respondents Indicating Poor or Worse Telephony Session Quality % with Good or Excellent Ratings % with good or excellent ratings Satisfied Some dissatisfaction High Dissatisfaction Telephony-enabled High Quality VoIP Packet Networksä G.729, 50ms delay, Policy, DiffServ, High fidelity 0% packet loss Extreme Dissatisfaction Round Trip Latency on Perceived Poor or Worse Round Trip Delay in msecs 0 Reduced ClearVoiceä Latency Today and Loss G.729, 70ms delay, 0% packet loss Typical Early Intranets Intranet Today G.723, 285ms delay, 3% packet loss (Selsius) Internet 2000 Internet Today G.723, G.721, 285ms delay, Reference: 5-10% packet loss Public Switched Telephone Network Today Source: Nortel Networks test of consumer expectations for voice quality, 1998 Source: VoIPMechanic UC Summit - PKE Consulting

11 Receiver Network Transmitter Time Model of VoIP One Way Latency Round trip time should be less than msecs or msecs one way (voice input to output) to meet criteria for telephony quality interactive voice - better results if kept under 95 msecs one way Frame X (In Input Buffer) Compression time = 3 msecs LAN Transmit Time Packetization time = 1 msecs Half Around the World Transmit Time Small Jitter Buffer 40 msecs this example Packet and decompression time = 3 msecs Frame 2 (In Jitter Buffer) Frame 1 Frame 2 (In Jitter Buffer) Frame Time in msecs UC Summit - PKE Consulting

12 Receiver Network Transmitter LAN Example of VoIP Example of a minimal latency LAN connection with G711 (no compression), 20 msec packets, and a 40 msec jitter buffer 60 msecs Frame X (In Input Buffer) LAN is almost instantaneous Small Jitter Buffer 30 msecs this example Frame 2 (In Jitter Buffer) Frame 1 (Output Buffer) Time in msecs UC Summit - PKE Consulting

13 Receiver Network Transmitter Impact of Multi-packet Jitter Buffer Example of a normal LD VoIP Call as Peer to Peer with 20 msec packets, compression, and a larger (60msec) jitter buffer Frame 6 (In Input Buffer) 125 msecs 2 frames in transit Network Time to Europe from US West Coast 3 Packet Jitter Buffer 60 msecs this example Frame 3 Frame 2 Frame Time in msecs UC Summit - PKE Consulting

14 Impact of Multi-packet Jitter Buffer and long distance transmission Receiver Network Transmitter Example of a VoIP call over an intercontinental IP link with a 60 msec jitter buffer Frame 8 (In Input Buffer) 175 msecs Network Time to India from US West Coast 4 Frames in transit 3 Packet Jitter Buffer 60 msecs this example Frame 3 Frame 2 Frame Time in Msecs UC Summit - PKE Consulting

15 The Multi-IP Packetization Issue TDM to TDM Local Loop 6-17 msecs total round trip latency 1-2 Msec round trip delay 5-15 Msec round trip delay PBX Local Loop TDM PSTN Results No need for Echo cancellation Latency does not cause collisions/interruptions TDM = 12.5 usecs per hop UC Summit - PKE Consulting

16 The Multi-IP Packetization Issue TDM to TDM Long Distance msecs total round trip latency plus LD transit time 2-3 Msec round trip delay msec Msec round trip delay Plus LD transit time PBX LD TDM PSTN Echo Canceller Results Echo cancellation required fixed time domain Latency does not cause collisions/interruptions UC Summit - PKE Consulting

17 The Multi-IP Packetization Issue VoIP to TDM msecs total round trip latency plus LD transit time Msec round trip delay msec Msec round trip delay Plus LD transit time Company LAN/WAN VoIP Gatewa y Echo Canceller TDM TDM PSTN Results Echo cancellation required fixed time domain Latency does not cause collisions/interruptions UC Summit - PKE Consulting

18 The Multi-IP Packetization Issue VoIP to VoIP - TDM PSTN core msecs total round trip latency plus LD transit time Msec round trip delay 5-10 msec Msec round trip delay Plus LD transit time Msec round trip delay Company LAN/WAN VoIP Gatewa y TDM TDM PSTN VoIP Gatewa y Company LAN/WAN Results Echo cancellation required variable time domain Latency may cause collisions/interruptions UC Summit - PKE Consulting

19 The Multi-IP Packetization Issue VoIP to VoIP - SIP PSTN core msecs total round trip latency plus IP Optical transmission time Msec round trip delay Msec round trip delay plus IP Optical transmission time Msec round trip delay Company LAN/WAN VoIP Gatewa y TDM VoIP Gatewa y SIP PSTN VoIP Gatewa y TDM VoIP Gatewa y Company LAN/WAN Results Echo cancellation required variable time domain Latency may cause collisions/interruptions UC Summit - PKE Consulting

20 The Multi-IP Packetization Issue VoIP to VoIP - Conferencing msecs total round trip latency plus IP Optical transmission time Msec round trip delay plus IP Optical transmission time Company LAN/WAN SIP IP Mixing Conference Bridge SIP PSTN Msec round trip delay plus IP Optical transmission time SIP Company LAN/WAN Results Echo cancellation required variable time domain Latency may cause collisions/interruptions UC Summit - PKE Consulting

21 Impact of SIP/ Buffering Regional IEC SIP Local SIP Vendor SIP IF / Company LAN/WAN Global Carrier SIP SIP 4 SIP Results Basic Latency msecs 14 hops??? Latency in each At 5 msecs/ average = msecs At 10 msecs/ average = msecs At 20 msecs/ average = msecs Plus Optical Tx time Global Carrier SIP SIP SIP Regional IEC Local SIP Vendor SIP IF / Company LAN/WAN UC Summit - PKE Consulting

22 Impact of Distance and Loops About 16 msecs each way About 79 msecs each way UC Summit - PKE Consulting

23 Impact of Distance and Loops Based on Where Service Providers Peer Can be 150 msecs or more each way About 42 msecs each way UC Summit - PKE Consulting

24 Impact of SIP trunking and Transcoding Global Carrier Global Carrier Regional 722 IEC SIP SIP SIP Regional 711 IEC SIP Local SIP 711Vendor SIP Local 722 SIP Vendor SIP IF / Company 722 LAN/WAN SIP Results Multiple Transcoding Quality degraded SIP SIP IF / Company 711 LAN/WAN UC Summit - PKE Consulting

25 The Impact is Like Recording Multiple Video Tape Copies Successive Encode/decode Processes with the Same Codec Algorithm UC Summit - PKE Consulting

26 Impact of Echo Echo comes from multiple sources TDM 2-4 wire analog conversions Crosstalk Devices Devices introduce echo Speaker to microphone coupling Heard by the other end UC Summit - PKE Consulting

27 Conferences are Obvious IP Mixing Conference Bridge Company LAN/WAN SIP PSTN Company LAN/WAN Results Multiple Echoes Variable Time Domains Echo TDM PSTN AnalogEcho Loop Echo UC Summit - PKE Consulting

28 The Result Postulate: 5-10% of VoIP to VoIP Calls that are NOT Peer to Peer have some Perceived Quality Issues UC Summit - PKE Consulting

29 Why is VoIP Quality an Issue Now? VoIP Adoption is passing 50% in North American Business (installed base) Business VoIP Adoption is 6-8% per year in the installed base (5% Churn and 1-3% upgrade) Home VoIP Adoption is increasing to 20-30% Carrier VoIP/SIP as the preferred backhaul is rapidly increasing VoLTE will make 250M cellular devices VoIP over the next 5 years UC Summit - PKE Consulting

30 Impact on an Individual VoIP Adopter UC Summit - PKE Consulting

31 VoIP Endpoint Adoption Impact on all Voice Calls UC Summit - PKE Consulting

32 Conferencing Impact Percentage of endpoints that are VoIP 10% 20% 30% 40% 50% 60% 70% 80% 90% 100% Percentage of Conferences with two or more VoIP Participants 5 Party Conferences 20% 80% 100% 100% 100% 100% 100% 100% 100% 100% 4 Party Conferences 9% 36% 81% 100% 100% 100% 100% 100% 100% 100% 3 Party Conferences 4% 16% 36% 64% 100% 100% 100% 100% 100% 100% Impact at 5% Bad IP Calls Percentage of endpoints that are VoIP 10% 20% 30% 40% 50% 60% 70% 80% 90% 100% Percentage of Conferences with Voice Quality Issues 5 Party Conferences 0.5% 2.0% 5% 8% 13% 18% 25% 32% 41% 50% 4 Party Conferences 0.2% 0.9% 2% 4% 6% 8% 11% 14% 18% 23% 3 Party Conferences 0.1% 0.4% 1% 2% 3% 4% 5% 6% 8% 10% Impact at 10% Bad IP Calls Percentage of endpoints that are VoIP 10% 20% 30% 40% 50% 60% 70% 80% 90% 100% Percentage of Conferences with Voice Quality Issues 5 Party Conferences 1.0% 4.0% 9% 16% 25% 36% 49% 64% 81% 100% 4 Party Conferences 0.5% 1.8% 4% 7% 11% 16% 22% 29% 36% 45% 3 Party Conferences 0.2% 0.8% 2% 3% 5% 7% 10% 13% 16% 20% UC Summit - PKE Consulting

33 UC Summit - PKE Consulting

34 Enterprises with a VoIP Premise Eliminate TDM s ASAP Eliminate internal loops (best exit routing based on IP path) Pick SIP trunking providers based on peering and paths Closer to the core and major IECs is best More connections is better Connections and operations based on eliminating loops Talk about encoding with your SIP provider What they do How often do they transcode Have they chosen peers based on transcoding Manage your LAN/WAN and VoIP environment well Potentially move to a cloud VoIP provider UC Summit - PKE Consulting

35 Service Providers Increase peering points to minimize loops and hops Choose paths based on latency, not cost Choose/modify encoding based on the pairing Negotiate encoding to minimize impact Analyze paths and routes for issues UC Summit - PKE Consulting

36 Industry Agree on a common encoding (or two) Low and high BW Negotiate for LCD to define coding and use end to end Develop a mechanism to use hops and loops to choose routing path in both SIP and IP Do entry echo cancellation in ALL VoIP endpoints Have standards for measurement and comparison Publish ratings Move to peer to peer implementations Open standards like WebRTC Publish open federation at the IP media level to enable peer to peer among vendors (Skype. SkypeB (Lync), Avaya, Cisco, etc.) Give us 10% of the Internet BW for real-time traffic UC Summit - PKE Consulting

37 For More Info PKE Consulting White Papers VoIPmageddon White Paper Making IP Networks VoIP Enabled CoS and QoS Managing bandwidth, Complexity, and Cost Defining VoIP/UC System Availability and Business Continuity No Jitter Posts VoIPmaggedon Please Mr. Chairman, Can't We Have 10% for Real-Time? UC Summit - PKE Consulting

38 Thank You Phil Edholm