Shunra Software

VoIP Testing

VoIP Testing

Ensure a Successful VoIP Deployment

Voice-over-IP (VoIP) delivers a wide range of benefits, including significant cost savings, and the management of one network instead of two. But combining voice, data and sometimes even video on the same corporate network is risky and requires careful planning. You must ensure that your network can support all three services running concurrently, and that they will each deliver acceptable quality while providing non-stop availability even if new applications are added or the network itself changes.

Shunra Virtual Enterprise (Shunra VE) for VoIP is a highly flexible and comprehensive network simulation solution. It creates an exact replica of your production network and remote end-user locations in your pre-deployment lab. It also integrates with and controls VoIP call generators, such as Ixia's IxChariot, IxVoice, and Spirent's Abacus. And you can even connect SIP phones directly to Shunra VE and hear first-hand how voice calls will sound over existing or projected network conditions. Shunra VE for VoIP delivers a powerful, complete and easy way to fully test VoIP performance well before you roll it out to production.

With Shunra VE for VoIP you will:

  • Know in advance that your network is 'VoIP ready'
  • Determine remote end-user call quality before rollout
  • Understand the impact of VoIP traffic on other mission critical data and video applications
  • Accurately determine the network capacity resources needed to support Triple Play
  • Optimize QoS and MPLS architecture infrastructures
  • Assess the impact of planned network changes on VoIP performance

Easily Plan & Test Your IP Convergence or Triple Play Implementations

Shunra VE for VoIP is ideal for accurately planning, testing and managing your IP convergence and triple play implementations. Using Shunra VE for VoIP you can create a near endless number of network architectures, and test data, voice and video applications side by side over each simulated network. You can even determine the right infrastructure and network capacity needed to deliver the required performance to all remote end-users.

Make Confident Decisions on VoIP Before You Buy

With Shunra VE you can accurately stress test a VoIP vendors' solution under any combination of network conditions, such as latency, packet loss, bandwidth, congestion, BER, link disconnects, and traffic conditions such as combined IPv4 and IPv6 traffic, triple play traffic, etc., and proactively determine call quality for all your remote locations – before you buy. You can also assess VoIP performance under future conditions – such as increased traffic or additional remote locations. Just as important, with Shunra VE you can keep on performing these tests periodically to make sure that changes in your enterprise environment won't negatively impact your critical voice services.

Real Voice Quality Analysis

Every company's corporate network is different, and a simple MOS measurement can't tell you what kind of voice quality your network can actually deliver at all remote locations. With Shunra VE you can actually hear the expected call quality from every remote location and even record the calls for future playback and analysis, during the testing process.

Deep Analysis and Comprehensive Reports

In addition to the empirical experience, Shunra VE for VoIP provides a wide range of graphically rich reports and drill-down analysis capabilities. These help you detect performance problems early, determine whether any modifications to the network or infrastructure are needed, and enable you to select the right equipment –– before you start your VoIP implementation.

Reports include:

  • Throughput and bandwidth utilization for capacity planning and verifying VoIP equipment performance
  • Voice quality in MOS and R values for each remote location, to determine the call quality from each remote location. Similar reports are available for Video traffic including MDI and VQFactor reports
  • Playback of recorded voice and video conversations
  • A SIP conversation report that measures performance during each and every stage of every call. This is ideal for troubleshooting problems

Supported Protocols:

SIP, H323, Skinny, MGCP, MEGACO, RTP, RTSP

Supported VoIP and Video Codex:

G.711, G.729, GSM, G.723, G.722, G.726, G.721, MPEG layer (I, II, III, IV) AMR (GSM, 3GPP), iLBC, ASF, H.264, JPEG, H.263, H.261

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