CSA Triple Play Analysis Suite - VoIP Page 1 of 5

Standard VoIP Protocol

Non-intrusive voice quality monitoring and evaluation based on the industry standard ITU-T G.107 E Model ( MOS and R-Factor )

RTP voice stream analysis in real time including RTP Jitter、Packet loss, Codec Type, SSRC identify and Session base throughput…etc., per industry standard RFC-1889

Option applicable for CSA’s ADSL, VDSL and GigE Platform Interface Modules (PIM)

VoIP SIP phone simulation in Peer to Peer or Proxy mode at either DSL loop or Ethernet interface

Support SIP based signaling analysis per RFC-3261 in Ladder Diagram, every SIP call/transaction is individually recorded and traced in graphic call flow to present all signaling point registered with IP address/port number, delta time and an alarm icon indicating the violation of SIP signaling

Most Voice over IP service providers are facing immense challenges which entail optimum network quality environment when rolling out Triple Play networks. AnaCise VoIP Triple Play Analysis Suite is the first complete solution in the industry, to run on a handheld CSA <Convergent Service Analyzer> emulating the VoIP SIP Phone or real-time monitoring for RTP traffic stream analysis. Basic features include display of specific call statistics, call signaling trace, and analysis to obtain QoS statistics, such as packet loss, jitter (Variations in delay of packet delivery) and MOS.The Triple Play Analysis Suite – VoIP, primarily tests and optimizes the Triple Play environment and with sophisticated yet simple to operate applications to assist every service provider during installation, testing, verification and troubleshooting of the VoIP service/s at client site.

The VoIP Triple Play Analysis Suite is focused onthe SIP environment and supports the VoIPprotocol stack based on SIP, G.711, RTP/RTCPand other protocols. Despite the protocol stack complexity, the VoIP Triple Play Analysis Suiteprovides a user-friendly interface, with easy touse tools for fast troubleshooting.

The Triple Play Analysis Suite - VoIP

Key Features

G.711

Ethernet / PPP

XDSL, Metro Ethernet

RTP / RTCP

UDPTCP

SIP

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Non-intrusive Sound Quality Testing PIM-XXX-STPA-VOIP

RTP Packet Analysis PIM-XXX-STPA-VOIP

VoIP Expert Analysis software employs MOS ITU-T G.107 and R-Factor to evaluate and rate thevoice telephone traffic quality and incorporates an extended version of the E Model analyzing theeffects of time varying IP network impairments which provides a more accurate estimate of useropinion. The sound quality test application’s flexibility and user-friendliness enable you to analyze the call quality in many ways such as non-intrusive at Ethernet interface or intrusive at DSL analog loop. It allows you to set up the analysis using 3 parameters, chosen between R-factor to Mean Opinion Score (MOS), Listener Quality (MOS-LQ), Mean Opinion Score PESQ (MOS-PQ), Conversational Quality (MOS-CQ), G.107 and R Conversation and evaluate those against each other.

Objectively evaluates the media quality of any live traffic-based calls according to the MOS scale(ITU-T P.800), based on the network industry standard ITU-T G.107 E Model (which provides R-Factor evaluation) and voice quality evaluation.

Objectively evaluates the media quality of any live traffic-based calls according to the MOS scale (ITU-T P.800), based on the network industry standard ITU-T G.107 E Model (which provides R-Factor evaluation) and voice quality evaluation.

No end-user can tolerate inferior VoIP phone call quality in a Triple Play network, as any packet delays may cause unpleasant echo and missing details in the conversation.Packets delays/loss results in conversation interruption while Jitter causes strange noises.

VoIP Expert Analysis software provides detailed RTP voice stream packet analysis of packet delay variation, packet loss, jitter, and other parameters in order establishing the benchmark and to aid the optimization process.

VoIP Call Quallity Analysis

DSLAMOr

L3 Switch

DSL ModemOr

Residential Gateway

Voice VideoData

CopperOr

Fiber OpticsBRASIPTV

NGN IP NET

NGN VoIP

ISP/ASPCSA

EthernetPassThrough

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SIP Proxy Registration Panel

SIP Phone Outgoing Call Panel

SIP Phone Incoming Call Panel

IP Phone Simulation PIM-XXX-STPA-VOIP-SIP

VoIP SIP Phone Simulation option provides the ability to conduct active test calls at either thetermination point of DSL analog loop or Ethernet interface which engineer can use this IP Phone to link back to an automated VoIP Interactive Response (IVR) directly to conduct the voicequality test readily.Parameters include all of SIP signaling、voice clarity and standard QoS metrics, or human speech call to obtain a sample QoE (Quality of Experience) result in voice clarity for service turn and on-demand troubleshooting purposes.

With SIP Phone Simulation, no physical IP test phone is needed at the customer site, and testing the customer IP phone is made easy by simply using Peer to Peer function to verify basic phone failure or poor VoIP quality caused by the actual physical handset.

DSLAMor

Metro EthernetNetwork

Copperor

Fiber OpticsBRAS

IPTV

NGN IP NET

NGN VoIP

ISP/ASP

TerminationPoint

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SIP Signaling Analysis PIM-XXX-STPA-VOIP

In additional to RTP stream and sound quality testing, the VoIP Expert Analysis option also provides instant SIP signaling analysis and transactions interaction in the call flow involving RTP, RTCP and SIP(RFC-3261) protocols being used in association with the VoIP call, a process with many possible steps involved.Just to track down any possible problems, it clearly displays all signaling in a call trace ladder format for easy troubleshooting of those signaling messages. All timestamps are displayed for resolving timing related issues. The call process is indicated using easy to understand icons that make it simple to follow all steps in the call. Both source and destination IP addresses of every telephone conversation and those transaction parameters involved in the call are time stamped with detailed analysis of the SIP protocol.

Support for SIP (RFC-3261) signaling. Associates signaling, voice streams per call.Provides clear flow of the signaling messages in ladder diagram.Identification of all signaling endpoints participating in the call.Provides accurate timestamps of all signaling messages.Measurement of call setup timing.Off-line Analysis with support for Sniffer and Ethereal file formats.

Multi-SIP Calls Detection and Monitor

SIP Signaling Flow In Ladder Diagram

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ANACISE TESTNOLOGY CORP.Fl. 3, No. 3, Alley 112, Ruei-Guang Rd., Neihu Dist., Taipei 114, Taiwan, R.O.C.

Tel : +886-2-2792-8880Fax : +886-2-2792-8058

E-mail : sales@anacise.comWeb : www.anacise.com

Note: Specifications subject to change without notice. All product and company names are trademarks of their respective corporations. © 2007 AnaCise Testnology Corp. All rights reserved.

Ordering InformationDo select appropriate part number/s of option/s within respective tables below with respect to the Platform Interface Module.Please contact local business partner or AnaCise for product configuration clarifications.

Convergent Service Analyzer

ADSL Triple Play PIMPIM-11A/B-ADSLA/B, PIM-12A/B-ADSLA/B and PIM-13A/B-ADSLA/B

PIM-1XX-STPA-VOIPADSL Triple Play Analysis Suite - VoIP Expert Analysis Option selected with EPT mode ONLY

PIM-1XX-STPA-VOIP-STBADSL Triple Play Analysis Suite – VoIP SIP Phone Simulation Option selected with MT/ET modes ONLY

GigE Triple Play PIMPIM-4XS-GigE, PIM-4XD-GigE

PIM-4XD-STPA-VOIPGigE Triple Play Analysis Suite - VoIP Expert Analysis Option selected with Dual Ports version

PIM-4XX-STPA-VOIP-STBGigE Triple Play Analysis Suite - VoIP SIP Phone Simulation Option selected with Single or Dual Port version