Most companies have firewalls, proxy servers, intrusion detection and/or web filtering devices in place. These mechanisms may require modifications to ensure a quality user experience for the rich media and interactive features of the platform.
The list below includes hostnames and ports used by the platform as well as the corresponding application functions they provide. If you are having issues with the interactive and/or rich media features of the platform, this information can be used by a network administrator to help resolve them.
This page is updated when necessary, so please bookmark it for future reference.
Function | Host Name(s) | Protocols/Ports | ||
Viewer Connection Information | ||||
Adaptive Bitrate Video Streaming Delivery (Live & VOD) |
Host Name(s) | MEDIA-LIVE.ONLINEXPERIENCES.COM MEDIA-VOD.ONLINEXPERIENCES.COM LEGACY.ONLINEXPERIENCES.COM |
Protocols/Ports | HTTPS 443 |
Static Content Delivery (Images, PPT Slide Content, etc)-PRIMARY | Host Name(s) | PRESENTATIONS.ONLINEXPERIENCES.COM MEDIA-ASSETS.ONLINEXPERIENCES.COM STATIC.ONLINEXPERIENCES.COM |
Protocols/Ports | HTTPS 443 |
General Platform Logon/HTML Wrapper (HTML, JS, Data Calls) | Host Name(s) | ONLINEXPERIENCES.COM | Protocols/Ports | HTTPS 443 |
Websockets | Host Name(s) | VC.ONLINEXPERIENCES.COM WEBSOC.ONLINEXPERIENCES.COM |
Protocols/Ports | HTTPS 443 |
Static Content Delivery (Images, PPT Slide Content, etc)-BACKUP | Host Name(s) | CONTENT.ONLINEXPERIENCES.COM | Protocols/Ports | HTTPS 443 |
Uploading Content & Assets | Host Name(s) | UPLOAD.ONLINEXPERIENCES.COM | Protocols/Ports | HTTPS 443 | Akamai Time Server | Host Name(s) | TIME.AKAMAI.COM | Protocols/Ports | HTTPS 443 |
Presenter / Producer Connection Information | ||||
WebRTC STUN binding - Presenter candidate gathering WebRTC DATA - Presenter Real-time Streaming WebRTC FALLBACK over Tunneled TCP |
Host Name(s) | *ING.ONLINEXPERIENCES.COM | Protocols/Ports | UDP 3478 (STUN) UDP 49152-65535 (DATA) TCP 80, 443 (FALLBACK) |
Encoder and Other Connection Information | ||||
Encoders "RTMP Based" | Host Name(s) | VC.ONLINEXPERIENCES.COM | Protocols/Ports | TCP 1935 |
VCU over SIP | Host Name(s) | VC.ONLINEXPERIENCES.COM | Protocols/Ports | HTTPS 443 SIP(TCP) 5060,5061 SIP(UDP) 5060,5061 |
Studio API Access | Host Name(s) | ONLINEXPERIENCES.COM | Protocols/Ports | HTTPS 443 |
As Needed | ||||
Mail Relay (only if email sender is custom and not @theonlinexpo.com) | Protocols/Ports | SMTP 25 |
For additional information that may assist your IT department with understanding the WebRTC protocol and how to successfully allow WebRTC streaming through a highly secure network, please consult this WebRTC knowledge base article:
https://blog.addpipe.com/troubleshooting-webrtc-connection-issues/
Also, it is possible to restrict the range of local UDP ports used for WebRTC in the Chrome browser with a registry setting:
https://chromeenterprise.google/policies/
Here is an overview of the network requirements for using Studio as a presenter. In this document you will learn more about the technology the platform uses to support real-time communications:
https://support.notified.com/hc/en-us/articles/115004317583-WebRTC-FAQ-