Tools

HTTP WebRTC client simulator - Used for HTTP, HTTPS, SOAP, REST automation / load testing.

It comes with in-built audio, video capability to enable testing of HTTP based multimedia applications.

Additionally it supports STUN, TURN and ICE to support WebRTC based product testing.

For audio / video testing, max concurrent media streams depends on bandwidth available. You can subscribe to dedicated bandwidth for higher number of calls.

SIP WebRTC client simulator - Used for SIP, IMS automation / load testing.

It comes with in-built audio, video capability to enable testing of SIP based multimedia applications.

Additionally it supports STUN, TURN and ICE to support WebRTC based product testing.

For audio / video testing, max concurrent media streams depends on bandwidth available. You can subscribe to dedicated bandwidth for higher number of calls.

APAT WebRTC client simulator - Used for proprietary automation / load testing. You can send / receive any text / binary messages using this module. For example, you can test DNS, DHCP, FTP etc using this module. Also you can use it to simulate JSON, XML, EBML messages.

It comes with in-built audio, video capability to enable testing of proprietary multimedia applications.

Additionally it supports STUN, TURN and ICE to support WebRTC based product testing.

For audio / video testing, max concurrent media streams depends on bandwidth available. You can subscribe to dedicated bandwidth for higher number of calls.

HTTP Content Server simulator - Used for simulating HTTP content servers. You can store images, audio / video files, data files etc and download from a content server. These kind of simulators are required if you want to generate huge traffic in network.

SIP Proxy simulator - Used for simulating SIP / IMS proxy servers.

It comes with in-built audio, video capability to enable testing of SIP multimedia applications.

Additionally it supports STUN, TURN and ICE to support WebRTC based product testing.

For audio / video testing, max concurrent media streams depends on bandwidth available. You can subscribe to dedicated bandwidth for higher number of calls.

APAT Server simulator - Used for simulating proprietary servers. It talks to all different kind of clients, plays a proxy role to route the messages between clients.

It comes with in-built audio, video capability to enable testing of proprietary multimedia applications.

Additionally it supports STUN, TURN and ICE to support WebRTC based product testing.

For audio / video testing, max concurrent media streams depends on bandwidth available. You can subscribe to dedicated bandwidth for higher number of calls.

JSON Proxy simulator - Used for simulating JSON based proxy servers. When you want to test your TURN server, you can make use of APAT WebRTC cleint + JSON proxy server combination and test your TURN server without involving your signalling server.