acterna voip 101 webinar may 05 nar

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Information about acterna voip 101 webinar may 05 nar
Product-Training-Manuals

Published on June 16, 2007

Author: Pumbaa

Source: authorstream.com

Acterna Presents:  VoIP Basics Acterna Presents Agenda:  Agenda What is VoIP ? VoIP Implementations VoIP Challenges/Impairments Test and Troubleshooting Acterna VoIP Suite Problem Solving Examples What is VoIP? - Break it down:  What is VoIP? - Break it down The transport of real-time interactive voice conversations using IP networks VoIP is Another IP application like e-mail and http A revenue generating service marketed to end users A cost savings Technology for the enterprise or carrier Signaling and Media:  Signaling and Media Signaling makes the connection Media Carries the voice Call Management Call Management VoIP – Signaling Protocols:  H.323 – The oldest and widely deployed Few new applications – Q.931 based SIP – The current favorite and most extensible IP phones, Centrex services, ATA’s, Soft Clients (Skype, X-ten) MGCP – Stateless – requires Gateway Controller NCS variant used in Cable TV MEGACO – Mix of the above Carrier Soft-Switch applications SCCP – Cisco proprietary IP Telephones, not normally used outside the Enterprise UniStim – Nortel proprietary IP Telephones, not normally used outside the Enterprise Which is Best? Pick the phone system that works for you VoIP – Signaling Protocols VoIP – Voice Encoding:  G.711 Same as used in current PSTN (Mu-law and A-law) 64Kbps - highest Bandwidth – lowest delay – Highest quality G.729A/B/C 8Kbps - low bandwidth – moderate delay – good quality Commonly deployed across the WAN G.723.1 5.3 or 6.3Kbps - lowest bandwidth – acceptable quality - longest delay Not widely used in VoIP VoIP – Voice Encoding VoIP - The Protocol Stack:  VoIP - The Protocol Stack Always Present in VoIP Implementation / Manufacturer Dependent OR Network Dependent VoIP – Packet Structure:  Real-Time Protocol Defined in RFC 3550 Used by H.323, SIP, MGCP/NCS, MEGACO and others Transports real-time interactive Voice and Video content VoIP – Packet Structure VoIP - Real Time Protocol (RTP):  VoIP - Real Time Protocol (RTP) What is VoIP?Bandwidth Usage:  Silence Suppression saves on bandwidth consumption Silence Suppression triggers Comfort Noise Point to Point WAN circuits may use Compressed RTP Reduces IP/UDP/RTP header from 40 bytes to 4 bytes What is VoIP? Bandwidth Usage What is VoIP?RTP and RTCP Operation:  RTCP quality monitoring packets are generated every 3 to 20 seconds Receive Packet counts, Jitter values, packet loss…. RTP RTP RTP RTP RTCP RTP packets will be generated every 10 to 30 milliseconds depending upon CODEC selection and configuration Not all Vendors implement RTCP but it can be useful because: Improved problem isolation and segmentation Supports calculation of Latency (Round Trip Time) Makes Possible single point analysis of end-to-end quality Provides more accurate MOS scores What is VoIP? RTP and RTCP Operation What is VoIP?Network Components:  H.323 Gatekeeper Phone number to IP lookup Registration, Authorization andamp; Security functions What is VoIP? Network Components SIP Servers Redirect server allows clients to signal directly Proxy server performs signaling on behalf of client MGCP / NCS Media Gateway controller MGCP mandates external Gateway control Media Gateway Controller makes all decisions Controls gateways/IP phones What is VoIP?Network Components:  IP Telephones Typically Ethernet (10/100 or wireless Hardware (desktop) or Softphone (PC software) What is VoIP? Network Components PBX Gateway Module Adds IP support to TDM PBX – Trunk and/or phones Existing phone support with the addition of IP support Session Border Controller Connects Islands of VoIP Allows VoIP to work with Firewalls and NATs May integrate Firewall and NAT functions What is VoIP?Network Components:  Soft Switch Interface with signaling components for gateway and switching control functions Separates function of switching and control What is VoIP? Network Components Signaling Gateway Interfaces different signaling systems (SS7 – SIP) Does not make decisions (like Soft Switch) Media Gateway Converts media between different networks (not signaling) For example G.729 VoIP to G.711 ISDN VoIP: Toll BypassPBX IP upgrade:  VoIP: Toll Bypass PBX IP upgrade Enterprise Site A IP Network (LAN) VoIP and Data WAN IP Phones Enterprise Site B Standard Phones Standard Phones IP PBX PBX Interoffice calls over IP network Signaling and Media Gateway Sites May be in different countries No Long distance or Termination fees for calls between Site A and B PBX upgraded to add VoIP VoIP: Enterprise to the DesktopIP PBX:  VoIP: Enterprise to the Desktop IP PBX Hedquarters Remote office IP Network (LAN) IP Network (LAN) IP Phones IP Phones VoIP and Data WAN ISDN Call management Signaling Control Router Router On Net: Source and destination are on the same network VoIP: Enterprise to the Desktop:  VoIP: Enterprise to the Desktop Hedquarters Remote office IP Network (LAN) IP Network (LAN) IP Phones IP Phones VoIP and Data WAN ISDN Call management Signaling Control Router Router Off Net: Source and destination are on different networks VoIP: Hosted (IP Centrex):  VoIP: Hosted (IP Centrex) Enterprise Remote Office Enterprise Head Office IP Network (LAN) IP Network (LAN) IP Network WAN IP Phones IP Phones VoIP Call Manager Router Router Service Provider Central Office VoIP: Virtual Phone ProvidersVonage SKYPE:  VoIP: Virtual Phone Providers Vonage SKYPE Home User Broadband Modem VoIP providers call manager Slide20:  Issues - Transport Related Slide21:  Issues - Transport Related Slide22:  Issues - Non-Transport Gateway 2 Wire analog IP Network Echo Reflection of speakers voice to speakers ear Not a true digital or VoIP problem Aggravated by network latency G.711 G.711 G.723.1 G.728 G.729 Issues - Human:  Issues - Human New Technology Lack of experience Lack of proven processes/procedures New technology requires new procedures Lack of resources Limited number of VoIP experts Subjectivity of complaints The sound quality is poor Slide24:  Testing - Out of Service PSQM – PESQ - PAMS Uses Prerecorded audio samples Compares receive audio to sent audio PESQ is expanded PSQM PAMS uses Analog input signal Router/Gateway Router/Gateway PSQM sample generator PSQM sample receiver Typically Ethernet Typically Ethernet Tools for Component Evaluation, Network Capacity testing Designed for Out-of-Service load generation for analysis of network components, features, capacity Not intended for In-Service monitoring, analysis and troubleshooting Slide25:  Testing - In Service Analysis MOS – Mean Opinion Score Quality based on Jitter/Packet loss etc.. Real-time or capture files of phone calls Relates to R factor and E model Router Router Tools for Service and Support of in Service VoIP Networks Designed for In-Service Monitoring of real user calls. Focus on troubleshooting customer problems Not for bulk call generation / simulation. Acterna VoIP Portfolio:  Acterna VoIP Portfolio Slide27:  HST-3000 Portable, battery powered Field Technician Tool Emulate IP telephone to place or answer VoIP calls Test VoIP signaling, connectivity and voice quality Gateway Call Manager Slide28:  HST-3000: Signaling Problems Non-functioning IP telephone Telephone cannot make or answer calls HST-3000 tests operation and configuration Emulate IP Phone Gateway Call Manager Test Call path Out of Service Test Slide29:  HST-3000: Voice Quality Poor Call Quality Customer dissatisfied with call quality HST-3000 performs quality analysis Emulate IP Phone Gateway Call Manager Test Call path Out of Service Test Slide30:  DA-3400 Network Analyzer Simultaneous VoIP and Data traffic analysis Real-time VoIP call quality monitor Real-time signaling monitoring and decoding Historical mode for identifying problems in the past DA-3400: VoIP Analysis:  DA-3400: VoIP Analysis Usage Mode: Long Term trending Real-Time monitoring DA-3400 Per Call Statistics:  DA-3400 Per Call Statistics Usage Mode: Detailed call stats Troubleshooting Slide33:  DA-3400: Issue Isolation Single Call Problem Solving Call experiencing poor quality Call Quality Screen identifies problems Call Problem Segmentation Call Quality Tab Slide34:  DA3400: Signaling Issues Call Signaling Analysis Problems with connecting using SIP Control Plane screen details signaling Control Plane Screen Tabs for each signaling protocol Select SIP Slide35:  DA-3400: Signaling Issues Call Signaling Analysis Problems with connecting Control Plane screen details signaling Filter on message type Filter on problem Station Address Message TAB Slide36:  PVA-1000 VoIP Analysis Suite Automated Remote Capture of VoIP calls Analysis of jitter, packet loss and call quality Audio playback with jitter buffer emulation Slide37:  Overview PVA-1000 Capture Agent Slide38:  PVA-1000 Capture Agent Remote Agent Capture of phone calls End user experiences poor quality call Selecting 'Save' sends call capture file to Support No need to send engineer to remote location Start Monitor indicator PVA-1000 Voice Analysis:  PVA-1000 Voice Analysis Audio playback with jitter buffer emulation, convert audio to WAV file, Jitter and Packet loss statistics Jitter and packet loss graphs Full protocol decodes PVA-1000 – Analysis Software:  PVA-1000 – Analysis Software Analysis and Playback features PVA-1000 – Analysis Software:  PVA-1000 – Analysis Software Jitter - Packet Loss Graphs Jitter Graph functions the same as the Packet Loss Graph Correlate measurable VoIP impairments to subjective audio quality. 'Hear what the user hears and 'see' the jitter and packet loss Slide42:  PVA-1000 with DA-3400 DA-3400 Console DA-3400 Auto-Captures on excessive utilization Analyze VoIP quality with PVA-1000 DA-3400 Console HST-3000 and PVA-1000working together:  HST-3000 and PVA-1000 working together Network Customer Prem HST-3000 Field Tech or phone Auto Answer Manual Answer NOC engineer uses PVA-1000 to: Analyze Jitter/Packet loss Playback Audio Decode Packets HST captures problem call HST forwards capture files to server Slide44:  Acterna VoIP Portfolio Problems SolvingExamples:  Problems Solving Examples Slide46:  Quality Cause and Effect Corporate Servers Headquarters Remote Office Converged VoIP and Data Networks Data traffic can impact voice quality Network Analyzers must be able to support VoIP and data simultaneously New Applications Server Moves Changes in number of workstations and phones Unstable Environment Changes in Traffic patterns and utilization 100% 10% Network Utilization Slide47:  Configuration Validation Verifying Packet Priority Settings VoIP should have proper priority Non-voice data should not have high priority Avoid competition for resources Conversation display shows stations and priority settings for each conversation, VoIP and Data Data Link Trend Slide48:  Intermittent Problems Broadcast Storms Intermittent burst of broadcast traffic Consumes network resources Impairs voice quality VoIP call quality summary VoIP Screen Poor quality trend aligns with broadcast storm Slide49:  Broadcast Storms Broadcast Storms Intermittent call quality problems Broadcast consumes network resources Impairs voice quality List all Expert Events Expert Screen Slide50:  Quality Notification SNMP – Email Notification Quality drops below threshold Users automatically notified SNMP and/or Email supported VoIP Setup Slide51:  Historic Analysis Looking Back in Time Quality issue happened 15 Minutes ago Select History Mode time boundaries Display only activity for selected time Select History Mode Define Time Period to display History Mode Slide52:  Call Quality Problem Single Call Problem Solving Call experiencing poor quality Identification of quality parameters needed VoIP Statistics Tab List statistics for all calls Filter on poor quality calls Slide53:  Problem Segmentation IP Telephones IP Telephones Endpoint A Endpoint B Poor Quality Call Call experiencing poor quality DA-3400 identifies problem segment Call Quality Tab Acterna Presents:  VoIP Basics Acterna Presents

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