Performer VoIP

Further product information.

The Performer is modular based, the individual modules handles call generation with objective quality assessment, load generation with H.323, SIP protocols, IP network simulation and detailed analysis of opened and closed calls in the network under test. The client/server concept allows the modules to be involved in a compact system and therefore forms testing scenarios according to user demands.

The Performer provides highly effective and accurate measurements offering user-friendly interface that enables an easy access to results and the ability to predefine user-defined tests. The exceptional features of the Performer dramatically reduce the time and costs connected with implementation and maintenance of VoIP. The key point is the Performer’s capability to capture, decode and analyze wide range of protocols. Part of the system is the Activity bar for signal detection after connecting to the channel. The complete VoIP testing suite includes SIPSim and 323Sim for load call generation, MediaPro for real time load monitoring and QPro for voice quality assessment in access networks.

The Performer’s software platform runs under MS Windows. There are two configurations available: portable PerformerLite or rack mounted system.

Description of the Performer’s test modules

QPro:

One of the essential parts of the Performer is the QPro, which performs objective measurements, analysis and voice quality prediction in the access networks on the same scale as subjective testing. The QPro enables call generation between itself and the other Performer’s components and also voice samples generation for objective voice quality measurements. The implemented algorithm compares the captured voice sample with corresponding reference and based on the mathematical model of human perception evaluates voice quality presented by MOS. The QPro offers link parameters measurements such as signal level, noise level, attenuation, level and latency of the reflected signal etc. and thereby speeds up the process of access network troubleshooting. Moreover the QPro features generation of calls with adjustable length for real network simulation, determination of signaling and call statistics and DTMF testing.

QPro’s Features:

• Voice Quality: PAMS and PESQ end-to-end quality testing
• Measures voice quality and GoS
• Simulates real-world conditions and stress
• The Cellular QPro: for cellular networks, supports protocols such as GSM, TDMA, CDMA, AMPS, NAMPS, EATCS, GPRS, UMTS

Optional Interfaces:

• 1 up to 8 analog ports
• Cellular QPro – runs over cellular networks
• 1 up to 4 ISDN BRI
• 1 up to 8 E1/T1 links (ISND, CAS, SS7 signalization)

Quality Call Analysis:

• Call Setup: Post-Dialing delay, Post-Gateway Answer Delay
• Voice Quality and Intelligibility: Listening Quality (MOS), Listening Effort (MOS)
• Signal Distortion: Background Noise, Signal Level, Insertion Loss, Echo, Echo delay, Return Loss, Latency, Round Trip Delay
• DTMF Signal Analysis, DTMF Integrity Testing, GoS evaluation

Stress Capability:

• The QPro simulates 240 simultaneous calls per module, 10% of which can be assigned as testing, the others as stress calls
• Unique Scalability of the Performer multi-topology architecture: enables 4QPro per Performer, up to 4 Performers can be controlled from one console, 5 consoles controlled by MasterScript
• Up to 30000 calls/hour generation per single Performer (1QPro)
• Adjustable length of calls: from 10sec to 10min
• Real network traffic simulation using three modules: Ramps-opens calls successively or simultaneously, Poisson-opens or Normal-opens according to Poisson and Normal distribution
• The stress capability enables validating, evaluating and optimizing the performance of the tested network components or tested system.
  

MediaPro:

The MediaPro is a cutting-edge, high-performance, session-oriented IP Multi-media analyzer (IPMM), provides accurate problem detection, fast troubleshooting and IP network performance analysis. Therefore the MediaPro represents an ideal tool for all VoIP users, ISPs, testing laboratories etc. The MediaPro analyzes a whole scale of parameters, which are important for ensuring the defined quality and grade of service in IP networks. Among such parameters are especially QoS, GoS and complete signaling, compliance with standards etc. What is more, the MediaPro presents a complete picture of the network status, displays opened and terminated calls. Supports IPMM signaling, including SIP, H.323, MGCP, Megaco, SCCP, NCS, TGCP, SIPT, C-SIP and more. The MediaPro enables detailed analysis not only of voice channels but also of all captured calls. The MediaPro ensures high quality analysis even under heavy stress. In addition, the MediaPro combines the objective voice quality evaluation functionality based on PAMS and PESQ, with subjective evaluation of voice quality based on a special playback mechanism.

MediaPro’s Features:

• on-line voice analysis in IP networks:

  • QoS a GoS determination
  • Standards compliance verification
  • Completeness of signalization verification
  • Supports SIP, H.323, MGCP, Megaco, SCCP, NCS, TGCP, SIPT, C-SIP signaling
  • Analyzes incoming calls from QPro, SIPSim and 323Sim
  • Displays statistics with full graphical support
  • Network statistics: jitter, packet loss, call activity and signaling, number of opened RTP streams
  • Call Activity: source and destination IP signaling, number of media and RTCP channels, jitter and packet loss in the signaling channel, possibility of on-line details about media channels, source and destination IP, jitter, packet loss analysis, G.723, G.729,…, H.263 and MPEG4 support
  • Opened and closed channels: source and destination IP signaling, jitter and packet loss, time stamp Call Setup, call duration and termination cause, protocol signaling and decoding of corresponding call, detailed statistics and information about specific RTP channels, voice decompression with user defined value of jitter, quality of voice evaluation using PAMS/PESQ.

• offline voice analysis in IP networks:

  • Provides an accurate timestamp for each call signaling, voice channels (RTP,RTCP) 
  • Complete statistics (GoS, Average jitter and packet loss).
  • Graphical format includes: call duration, bandwidth utilization, distribution and cause of faults , packet loss

The HW equipment which is exploited by the MediaPro can be shared also with another SW application: the Capture Protocol Analyzer supports decoding of more than 450 protocols and provides detailed statistics of captured traffic.

SIPSim

The SIPSim,a Services over IP (SOIP) Performer component, is an advanced hardware and software solution that generates high volume SIP-based services for SoIP systems testing, combining both signaling and media. By emulating a large number of SIP terminals that can initiate one or more SIP sessions, receive network SIP responses and terminate existing calls, the SIPSim is capable of stressing different network elements such as Proxy servers, Registration servers, Redirect servers and Application servers. Additionally, RTP voice packets can be generated, supporting different codecs. The SIPSim emulates real world conditions and enables developers, QA labs and Service Providers to benchmark, load test and verify proper protocol implementation in VoIP equipment, as well as ascertain compliance to standards, signaling integrity and Grade of Service.

Features:

• Generation of more than 200 calls per second
• Generation of up to 2000 SIP calls simultaneously
• Generation of up to 300 RTP channels simultaneously
• Supports codecs according to G.723.1, G729A, G.711, G.722, G.726, G.728, H.263 and MPEG-4
• Enables to set priority (emergency, urgent, normal a non-urgent)

323Sim

The 323Sim is a high performance H.323-based VoIP services simulator, enables developers and service providers to benchmark, load test and verify proper protocol implementation in H.323-compliant, Voice and Video over IP equipment. The 323Sim can test gateways, gatekeepers and other VoIP products and is capable of generating thousands of simultaneous calls (over 80 000 calls to Performer per hour) with the ability to control and configure call parameters. The 323Sim supports transmission of RTP packets, encoded in various codec formats. Enables activation of specific portions of the call setup process: RAS, Q.931 a H.245. Displays call activity and completion statistics. Additionally supports custom scripts creation functionality for automated tests.

323Sim Features:

• Generation and reception of more than 80 000 calls per hour
• Configuration of RAS, Q.931 a H.245
• Supports codecs according to G.723.1, G.711, G.722 a G.729
• Supports H.261, H.263 and MPEG-4 video streams
• Provides DMTF integrity testing

MasterScript

The MasterScript represents a powerful SW tool, which enables individual Performer modules to be programmed from central console and furthermore enables the creation of user test profiles including stress call generation, voice quality analysis and results processing using user-defined format.

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