Audio Conferencing IP and TDM May 5, 2004 Jerry Norton Chief Technical Officer Vapps, Inc. Wendell Bishop Chief Scientist NMS Communications
NMS at a Glance Founded in 1983, publicly traded since 1994 2003 revenues of $87 million 300+ employees 130+ in R&D Designed into products deployed in 90 countries Major telecom operators, equipment and solution providers rely on NMS Product Development Sales & SE Support Channel Partner s Headquarters Slide 4
The NMS Business Operators, Equipment Manufacturers, Solution Providers Worldwide Sales / Distribution Partners Communication Building Blocks: Enabling technology, systembuilding blocks, and complete systems Voice Quality: Standalone systems, embedded solutions Wireless Access Gateways: 2G & 3G infrastructure NMS Technology and Service Leadership Application and Technology Partners Best-in in-class Supply Chain Strong Financial Position Slide 5
Diverse Applications Video, audio & web conferencing Multimodal and messaging applications Unified communications Voice Security Wireless video streaming Personalized caller entertainment Media and voice application servers Speech-driven applications Business communication and contact centers Slide 6
Audio Conferencing Wendell Bishop Chief Scientist NMS Communications
Audio Conferencing Subsystems Last month new conferencing applications This month audio conferencing only Drill down to technical aspects of subsystems Next month focus on video conferencing Don t make assumptions about nomenclature Overloaded terms unavoidable Hereafter conferencing audio conferencing While NMS has audio conferencing products that can be used in building anything discussed in this presentation, individual product features will not be discussed in detail - contact me for further information Slide 8
Hierarchy of System Elements Focus is on the conference-specific components Not all systems use all four levels Conference Application Conferencing Services Conference Engine DSP Filters Slide 9
DSP Filters Mixer, vocoders, talk detectors (Echo cancel, DTMF detect/suppress, play, record) Well defined single functions no decisions Applied to audio stream connected from above Mixer + Listener only output Talker 1 input Talker 2 input Talker 3 input - - - Talker 1 output Talker 2 output Talker 3 output Slide 10
Conference Engine Single talk domain per conference All members listening to the same audio content Internal dynamic routing of active member audio Active talker decisions based on talk detection, priority Manages resources shared by many conferences Talker Listener + Talker Listener Listener Talk domain Slide 11 Listener
Conferencing Services Reconfiguration during a conference Add and remove members and talk domains Change talker permissions and priority Multiple talk domains sidebars, coaching IVR functions before and within conference Main member Conference recording Main conference talk domain Sidebar link Member moving to sidebar Sidebar talk domain Sidebar prompt Sidebar member Slide 12
Conference Application Handling call control as well as media streams Admission policies implemented IVR front-end script prior to entering conference Conference ID and passwords Conference management interface Reservation (resources, ID, passwords) Moderation assigning talker permissions / priorities Service environment integration Call detail records for billing Network management interface Slide 13
Applications Use Different Levels Instant Messaging or Group Calling Call center support system Commercial conferencing service Conference Application Conferencing Services Conference Engine DSP Filters Slide 14
Circuit to Packet Migration: DSP Filters Circuit time driven Mixer and talk detector DSP processor or special silicon 8KHz PCM (G.711) speech sample processing Packet data driven Mixer, talk detector, vocoders DSP processor or host processor Frame (block) processing Coded and higher quality speech possible Slide 15
Circuit to Packet Migration: Conference Engine Circuit switching Timeslot switching of TDM streams; sample by sample Single speech type (G.711 PCM; ADPCM possible) Generally, mixer is always in use Packet routing Speech moved in larger packets (frames) Mix of multiple speech types is possible Single talker may be forwarded bypassing mixer Especially on coded speech; avoid multiple vocodings Slide 16
Circuit to Packet Migration: Conference Services/Application Circuit audio centric PSTN call control IVR front end only for admission IVR and local LAN management Packet richer media SIP call control IVR and web based admission and management Coupling to other conference media Web broadcast supported Slide 17
Circuit Centric Mixed System Conference engine may / may not include GW GW involves aggregation delay / jitter buffering Note multiple encoding for G.729 members G.729 GW RTP G.729 members G.729 GW G.729 GW Circuit mixer PSTN G.711 members Always mixing Always vocoding Slide 18
Packet-Centric Mixed System Conference engine may / may not include GW GW involves aggregation delay / jitter buffering Minimal resource usage when single talker Single talk domain RTP G.729 members G.729 packet domain Linear domain & mixer G.711 packet domain G.711 GW G.711 GW G.711 GW PSTN G.711 members Mixing only as needed Vocoding only as needed Forwarding when single talker Slide 19
Talker Control Algorithm Tradeoffs Maximum number simultaneously mixed Good barge-in response among conversing members Efficient resource usage with mixing limits Single talker forwarding versus always mixing Forwarding low delay; less vocoding for better quality Always mixing instantaneous barge-in response Explicit versus implicit priority / permission Implicit different priorities fixed during conference Barge-in priority easy to manage up front Explicit listen only / talker changed during conference Moderated floor leads to more civil discussion Slide 20
Scalability Issues for: Individual Large Conferences Problem for lower levels of conference hierarchy Mixers of limited size can be cascaded But limit to broad and shallow for quality two levels Talk detection on all parties across many nodes Minimal resources for only talk detector on silent member Quickly move newly active talker to mixer node Requires good private circuit switching / packet routing Segregate listen-only members for broadcast Web broadcast for listen-only Streaming media delay is acceptable Slide 21
Scalability Issues for: Systems with Many Conferences Problem for higher levels of conference hierarchy Replication of standard DSP filters / conference engines Management of a multi-server system is needed Load balancing; interaction with network call control Handling out of service nodes; fault tolerance Moving callers between system nodes is needed Conference location discovered only after IVR interaction Front end hard switch, soft switch, or NAT device helps Slide 22
Developer / User Interfaces DSP filters Enable/disable (create/destroy) filters; connect streams Command and event message interaction Developer participates in active talker dynamic rerouting Conference engine Create/destroy conferences; add/remove members Change member priority; active talker events Special member types for higher level features Play, record, inter-conference link Developer participates in reconfiguration only Slide 23
Developer / User Interfaces Conferencing services May be part of larger media server system Support for integral play, record, multiple talk domains OAM monitoring and statistics Emerging conference control standard protocols Conference application Interacts with media server / call control server (softswitch) User friendly admission, management, and moderation IVR scripts tailored to type of participant GUI web interface Slide 24
Conference Bridge Design Real World Example VAPPS CB1000 Jerry Norton Chief Technical Officer Vapps, Inc.
VAPPS at a Glance Founded May 2002 to create a VoIP Conference Bridge Product commercially available since September 2003 Industry leading VoIP conference bridge for service providers and large enterprises. Customers include Conference Service Providers and resellers Dedicated Conferencing Access Conferencing Bow Communications Slide 26
Decision Process in Building a Conferencing Bridge Packet / TDM versus Packet only Multiple codecs versus single Scalability where and when to switch calls Target Market Changes Application Conference Service Provider End User (Corporate) Feature List Slide 27
Circuit or Packet? Conference Services/Application Circuit - existing customer base Embedded vendors Features easily duplicated High comfort level Packet - disruptive technology Logical rather than physical switching Lower cost of entry to new players 1/100 th cost of switching Slide 28
Circuit or Packet: Need to do both? Circuit - switching Easily converted to Voice over IP Core Network of Carriers changing to IP Delay time slightly better Switching matrix Circuits require TimeSlot Interchange Always blocking at some point On a particular time slot throughout call Slide 29
Codec Complexity: How much is worth the cost? Cost to load multiple codecs Complicated codecs use more MIPS than G711 Create different delays for different users Single talk domain RTP G.729 members G.729 packet domain Linear domain & mixer G.711 packet domain G.711 G.711 G.711 Mixing only as needed Delay to here longer than 711 MIPS and memory double Slide 30
Codec Complexity: Need to support? Complex codecs normally used to conserve bandwidth across Internet or Private Networks G723 better at interpolating dropped packets G711 does nothing VoIP service offerings (when they really exist) are all available G711, sometimes G729/726/723 Decision: include support, but optimize for G711 Slide 31
Scalability Issues DSP code handles voice detection/mixing Application determines how and where to put conferences Switching occurs on a single board, across boards, across chassis Option to use TDM switching even in packet environment Option to use VoIP redirection Slide 32
Scalability Issue: Routing Call On majority of calls, you do not know which conference it needs to join Using DNIS to put on correct board/chassis creates blocking on TDM bridges DNIS in VoIP can efficiently put call on correct board/chassis Using many DNIS entries reduces overall efficiency Slide 33
Scalability Issues: Options One board to another can use private bus Switching done via TSI eventual blocking VOIP can round robin the call requests SIP has inherent redirect method in process call can be sent to new media connection (move the call realtime) Cascading bridges across boards Cascading across chassis Slide 34
Scalability Options: Tradeoff Call enters and is placed on least used board When correct conference determined, switch If (on same chassis) and (loading < x) switch across TDM bus If (on different chassis) and (target loading < x) switch via VoIP methods Else: Cascade across Chassis Slide 35
System Elements target by market Conference Service Providers High End Corporate Networks API to existing controls Conference Application Conferencing Services Conference Engine DSP Filters Ease of use for Enduser and Admin Slide 36
Conference Service Provider Likely: has existing database of users Likely: has minimum required features Website for customer signup, another for customer s conference control and usage info Integrates information into existing environment Reduce price per conference port OAM features are very important Slide 37
Enterprise Conference Application More local use integrate with MS Outlook Smaller scale, large conferences less important Multiple office locations with duplicated environment Central Administrative control Security policies set by IT Slide 38
Feature list tradeoffs IVR Avoiding a busy tone or timeout IVR peaks within 1-2 minutes of each busy hour Less resources mean caller waits longer Rule of thumb: 1:10 ratio Multiple small boards with conference/packet handling and IVR give better performance Slide 39
Feature List Tradeoff Reservationless vs. Scheduled Prevent No resource available or trust to good luck Efficiency and blocking in TDM bridges led to need Massive Dial-Out for the Executive touch Switching capabilities eliminate resource issue Software layering allows easy introduction of scripted features compared to Big Iron Slide 40
Feature List Tradeoffs Features like sidebar and mentoring Used most often with other applications (call center or distance learning) Execute from web control or from DTMF Complexity versus incremental usage Features are implemented in DSP via Application coding Easy coding if DSP supports the feature If DSP doesn t support, uphill battle Slide 41
Thank You! Slide 42
Please take a moment now to complete our short survey, while we start the Q & A Slide 43
For more information Contact Wendell Bishop Jerry Norton + 508 271 1313 wendell_bishop@nmss.com + 917 861 7967 jerry@vappsinc.com June The Coming of Age of Video Conferencing Register now at: http://www.e-conference.com/nms1/springseries/ Take the Application Challenge! Visit http:///news/oachallenge.html and enter to demo your application at SUPERCOMM 2004 Slide 44
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