DiversifEye-VoIP test

Voice over IP (VoIP) is a key element of the Triple Play service. The integration of VoIP into a converged network requires performance testing on a per call basis with increased emphasis on the quality of experience for end-to-end connections.

VoIP Environment VoIP Quality of Experience

To ensure VoIP performance and quality, network delay must be insignificant. Packet latency or delay should not exceed 250ms, after this point sound quality will be perceived as poor. Ensuring QoS settings are correct on the network elements will help, however other components such as choice of codecs are important.

A mixture of call origin devices will mean several different codecs will be used in transmission of voice over the network. Traditional based testing is no longer feasible, a real world typical scenario is a GSM client calling a PC based soft phone. Quality on a per client flow basis will require in depth analysis at a protocol level (e.g. SIP and RTP) and codec level (e.g. GSM and G.711).

diversifEye™, the award winning test solution for converged IP networks enables service providers and equipment manufacturers to test end-end network and system performance. diversifEye™ tests include determining the number of call attempts capable during peak hours, determining root cause and affect on quality when integrating different call origin devices or codecs.

Key to the success of diversifEye™ is its per flow architecture which provides test results on a per client basis. As part of the performance analysis package are the active and passive metrics of MOS voice quality scoring including PESQ and R factor along with packet delay, loss and jitter analysis. Predictive quality analysis at packet level enables service providers to identify key elements that are causing degradation in their network environments.

VoIP Test Scenarios

VoIP quality in Triple Play service- Access network test to support Triple Play (DSLAM/FTTH/CMTS, Residential Gateways, BRAS etc.). Determine VoIP performance under converged IP scenarios, determine threshold performance by increasing VoIP and other IP applications to a point where packet loss effects call quality and establishment rates. SIP Server Call Rate - Emulate and analyze the call rate of the SIP server, with thousands of VoIP users on the client side. Clients are unable to send or receive RTP media on half duplex connections. Reassess the call rate by allowing RTP media and full duplex connections.