VoIP Load Tester

Description

H.323 and SIP are standards that define the protocols for audio-visual communication sessions over IP. They are both widely implemented by voice and videoconferencing equipment manufacturers.

The Valid8 VoIP Load Tester allows you to test Endpoints, PBXs Gateways, and test load and feature interaction for audio and video. It enables continuous route testing through the network to report quality and find issues. It is useful for small and large carriers: ILECs, CLECs, ASPs, ISPs, as well as enterprise customers. It automates deterministic verification of routes and phone numbers across VoIP networks, saving time and increasing effectiveness and coverage of testing. Rules are set up in the tool to cycle through multiple network destinations and phone numbers according to the test plan.

Features of our VoIP Load Tester

• Real-world device simulation supporting popular vendor modes and profiles
• Simultaneous SIP and H.323, WebRTC, H.264 and H.263, VP8, G.7xx, OPUS
• Measure Key Performance Indicators (KPIs) such as number of simultaneous sessions and Busy Hour Call Attempts (BHCA)
• Tests up to 5,000 sessions (calls) in parallel per core (scalable)
• ICE/TURN/STUN
• DTLS/SDES
• Custom REST/HTTP/JSON signaling
• Check dialing in/out of HD video systems
• Supports dialing through IVR using DTMF prior to activating video and audio playback
• Report on media received, call connect time,call duration, jitter, packet loss to check SLAs are being met
• Alerts and notifications
• Check parameters in messages from SUT and flag errors
• Generate valid and invalid/negative messages and call-scenarios, including malformed, dropped and misordered packets
• Continuously test routes and cycle through destinations across VoIP networks and call numbers according to configurable rules
• Check billing accuracy in conjunction with Radius/Diameter (optional)

What can it do?

The VoIP Load Tester is capable of simulating and testing several devices individually or in parallel. It can simulate WebRTC clients and test SBCs, ESBCs, proxies, and endpoints.

Media Performance 128kps to 1152kbps 1080p, 720p, 480p, 360p

15/30/60fps (frames per second)

Sample Test Scenarios

• Outgoing call
• Incoming call
• Standard video and audio call
• Standard video audio and presentation
• Start a call with audio only, then add video

Audio, Video validation

• Video bitrate
• Media detection during call
• SDP port
• Payload type
• Media type
• Basic Conformance Check (correctness of signaling parameters) Screenshare (future planned)

Performance

• Concurrent calls: Up to 500 (scalable)

Counters and Analytics Call Attempts

• Call Successes
• Call Failures Registration Attempts Registration Successes Registration Failures 3xx Failures
• 4xx Failures 5xx Failures
• Measurements
• Calls per second (CPS)
• Call setup time
• Call tear down time
• Media Tx Packets (audio/video)
• Media Rx Packets (audio/video)
• Media Jitter, Loss, Delay

Application Standards

• IETF RFC3261 – SIP
• ITU-T – H.323, H.225, H.245 slow start and fast start IETF RFC6455 – The WebSocket Protocol
• IETF draft-ietf-sipcore-sip-websocket IETF RFC2617 – HTTP Authentication IETF RFC2976 – SIP INFO
• IETF RFC3310 – AKA Authentication IETF RFC3311 – SIP UPDATE
• IETF RFC3428 – SIP MESSAGE IETF RFC3966 – tel URI
• IETF RFC4627 – JSON socket.io emulation

Transport Standards

• IETF RFC4347 – DTLS
• IETF RFC5246 – The Transport Layer Security (TLS) Protocol IETF RFC793 – Transmission Control Protocol (TCP)
• IETF RFC768 – User Datagram Protocol (UDP)
• IETF RFC4960 – Stream Control Transmission Protocol (SCTP) IETF RFC6455 – The WebSocket Protocol
• IETF draft-ietf-sipcore-sip-websocket

Network Standards

• RFC 791 – Internet Protocol, Version 6 (IPv4)
• RFC 2460 – Internet Protocol, Version 6 (IPv6)
• Media Standards IETF RFC2327 – SDP
• IETF RFC3264 – SDP Offer Answer IETF RFC3266 – SDP Ipv6 support IETF RFC3550 – RTP / RTCP
• IETF RFC3711 – SRTP
• IETF RFC5245 – Interactive Connectivity Establishment (ICE) IETF RFC5389 – Session Traversal Utilities for NAT (STUN) IETF RFC5766 – Traversal Using Relays around NAT (TURN) draft-perkins avt-rtp-and-rtcp-mux
• H.261, H.263, H.263+, H.264, VP8, H.264 Annex G SVC (future planned), H.265 (future planned)
• G.711 (u-law and A-law), G.722.1c (48kbps, 32kbps, 24kbps), G.722.1 (32kbps, 24kbps), G.722, G.729, OPUS
• DTMF (in-band and RFC2833)

PRODUCT DETAILS