This article is only applicable for those having audio issues such as one way speech or bad call quality when using softphone. If you experience other connection issues such as busy, offline or logon errors please read the Softphone Basic Information article. Please check information under the heading “Increasing debug level and sending log files to Puzzel” on how to send logs.
If you experience one way speech or other bad call quality issues, first go though the Softphone section in our Basic Requirement document. It might be that some required settings in your network are causing the issues.
If you have confirmed that your settings are done correct Puzzel Support can assist with the troubleshooting.
When contacting us we would appreciate if you can collect some additional logs.
- Please ask the affected agents to open a new Google Chrome tab and enter the following address: “chrome://webrtc-internals/”. Preferably beside the Puzzel Agent web page. The “chrome://webrtc-internals” is an internal Google Chrome tab that holds statistics about ongoing Softphone calls.
- When a call with the audio issues has ended switch back to the WebRTC Internals tab and click on the following:
- When the “Create Dump” line is pushed you will see a new button appear on the screen:
- Please press “Download the PeerConnection updates and stats data”. Make sure that the active tab is the one marked in this next picture. If you had several calls the last call will be the tab to the far right:
- After pushing the “Download the PeerConnection updates and stats data” button a file with the name “webrtc_internals_dump.txt” will appear in your download folder on your computer. Please transfer this file to Puzzels Help Center in an existing ticket or when raising a new one.
PLEASE NOTE: Because of the Chrome settings the agent has limited time to fetch this data for us.
- This last point is for those of you who would like to understand this tool in a more technical way and if you want to troubleshoot in your own environment. For Puzzel this information would be very useful so we can see the session stats also in graphic style. However this is optional for you. At the following picture you can see a number of tabs, one for all getUserMedia calls and one tab for each RTCPeerConnection.
On the GetUserMedia Requests tab we can see each call to getUserMedia.
For RTCPeerConnection stats we can see four things here:
- How the RTCPeerConnection was configured, i.e. what STUN and TURN servers are used and what options are set
- A trace of the PeerConnection API calls on the left side. These API traces show all the calls to the RTCPeerConnection object and their arguments (e.g. createOffer) as well as the callbacks and event emitters like onicecandidate. Oneicecandidate is the IP adresses used for Media connections. The ICE protocol will select one of the IP adresses from those candidates.
- The statistics gathered from the getStats() API on the right side
- Graphs generated from the getStats() API at the bottom. Make sure the agent has pushed the two lines marked in yellow “Stats graphs for Conn-audio” and “Stats graphs for ssrc_num_send (audio). In Google Chrome the agent will probably have to use the zoom function to get as much clear stats as possible.