Overview

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. It reports on media received, call connect time, call duration, jitter, packet loss. It verifies equipment is working and configured as it should and helps check that SLAs are being met.

Features

  • 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)

  • L4 Test Suite (optional)

VoIP Load Tester

What can it do?

The VoIP Load Tester solution is capable of simulating and testing several devices individually or in parallel:

It can simulate

UAC

UAS

Endpoints

It can test

Endpoints (audio and video)

Conference Bridges

Softswitches

Proxies

SBCs

Gateways (SIP to H.323)

Specifications

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

IETF RFC4975 - MSRP

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)

Network Standards

RFC 791 - Internet Protocol, Version 4 (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, 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, AMR NW/WB

DTMF (in-band and RFC2833)

Media Performance

128kps to 1152kbps

1080p, 720p, 480p, 360p

15/30/60fps (frames per second)

Presentation Sharing Standards and Protocols

H.264, H.263

Dual-video (BFCP, H.239)

Network Emulation

Simulated network delays and packet loss

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

- H.264 sequence parameter

- Video bitrate

- Media detection during call

- SDP port

- Payload type

- Media type

Registration (MD5, AKA Authentication)

Basic Conformance Check (To-tag, From-tag, Via branch, Via rport, SDP length, Max-forwards)

Performance

Concurrent calls: Up to 5,000 (Mini), 150 (Micro)

Calls per second (CPS): Up to 1,000

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

Additional Details

Options

VoIP Load Tester (150)

VoIP Load Tester (1,000)

VoIP Load Tester (5,000)

Related Solutions

Conformance Testing

Contact

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Copyright 2013-2017. All Rights Reserved

Specifications subject to change at any time