This page is provided as a simple way to evaluate whether a facility network allows access to the ZipLine 3.0 web phone.
At a computer inside the facility network use either Chrome or Firefox to make a test call to our Wideband Audio demo. The following button will launch ZipLine 3.0 web phone in a separate window.
|Launch a test call!||If you're using Chrome at this very moment you can try this for yourself right now! Just click on the big blue button to launch a test call.
If the browser connects successfully a greeting will be heard. This confirms that the network allows SIP traffic over TCP. This is all that is required for an interpreter using Zipline 3.0.
If the browser fails to connect the network firewall or router is blocking SIP traffic. Please ensure that the following ports are enabled in the router or firewall:
- SIP signaling: ports 5060-5070 (TCP & UDP)
- RTP media: ports 12000-21999 (RTP, TCP & UDP)
- DNS: port 53*
- NTP: port 123*
- HTTP: port 80*
- HTTPS: port 443*
*These ports are commonly allowed on any network that allows internet access.
|Q: Which direction would the ports need to be opened|
|A: Packets are going to flow in both directions.|
|Q: Who is initiating the connection|
|A: The webphone, running in the browser, will initiate the packet flow, and the server will respond once it receives the first packets.|
|Q: What are the destination IP’s|
|Q: What else is hosted at the destination IP’s|
|A: Everything running at those IP’s is operated by ZipDX LLC. That includes our conferencing and collaboration business, conversational texting, our email, plus specialty tools for the telecom industry.|
|Q: Who/How is the initiation of the connection being made|
|A: Conference participants (including interpreters) initiate the connection from the web browser using the WebRTC protocol.|