Hi everyone,
I’ve set up a SIP-based VoIP testing service on the AREDN network. It can be used to quickly test phone connectivity, audio quality, latency, codecs, and two-way RTP without registering the phone to a PBX.
Echo test
Dial 600@10.101.64.184 or Dial 10.101.64.184 and press 1.
Anything you say will be returned to you in real time. This is useful for testing microphone audio, speaker audio, latency, packet loss, and two-way RTP.
Audio playback test and client info
Dial 601@10.101.64.184 or Dial 10.101.64.184 and press 2.
The service responds by reading back the caller's SIP and VoIP client information.
Feedback and test reports are welcome!
I’ve set up a SIP-based VoIP testing service on the AREDN network. It can be used to quickly test phone connectivity, audio quality, latency, codecs, and two-way RTP without registering the phone to a PBX.
Echo test
Dial 600@10.101.64.184 or Dial 10.101.64.184 and press 1.
Anything you say will be returned to you in real time. This is useful for testing microphone audio, speaker audio, latency, packet loss, and two-way RTP.
Audio playback test and client info
Dial 601@10.101.64.184 or Dial 10.101.64.184 and press 2.
The service responds by reading back the caller's SIP and VoIP client information.
Feedback and test reports are welcome!