Overview

WebRTC (Web Real-Time Communication) is an open standard for real-time, plugin-free video, audio and data communication.

The Valid8 WebRTC Load Tester is capable of simulating WebRTC clients to test WebRTC proxies replicating conditions that are hard to manage manually.

Features

  • Secure (WSS) and plain-text (WS)

  • ICE/TURN/STUN

  • DTLS/SDES

  • Custom REST/HTTP/JSON signaling

  • Client-side JavaScript code can be imported and used directly in test scenarios

  • Supports SIP over W3C WebSocket

  • Supports custom APIs for call control including REST/HTTP, native JSON support

  • Measure Key Performance Indicators (KPIs) such as number of simultaneous sessions and Busy Hour Call Attempts (BHCA)

  • Report on media received, call connect time, call duration, jitter, packet loss to check SLAs are being met

  • Alerts and notifications

  • Mobile network impairments

  • 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)

WebRTC Load Tester

What can it do?

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

It can simulate

WebRTC clients

It can test

SBC

ESBC

Proxies

Endpoints

Specifications

Application Standards

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)

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

DTMF (in-band and RFC2833)

Video Performance

128kps to 1152kbps

1080p, 720p, 480p, 360p

15/30/60fps (frames per second)

Network Emulation

Simulated network delays and packet loss

Mobile network impairments

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)

Performance

Concurrent calls: Up to 500

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 (10 - 50)

VoIP Load Tester (50 - 500)

Contact

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Copyright 2013-2017. All Rights Reserved

Specifications subject to change at any time