WebRTC JsSIP Expert /needed For Debugging SIP VoIP Soft-Phone DTMF Issues

Job ID: 35160069

Budget: $30 – $250 USD

Hello,

We have a VoIP SIP Soft-Phone that uses WebRTC & JsSIP, our dial pad is not working correctly to enter extension #'s for the DTMF tones, also the audio quality is not the best and we need to fix that too, in the past it was a lot better, not sure why the change, the hard phones do not have these issues, just the soft-phones.

We need an expert who can solve these issues, please do not waste our time if you are not an WebRTC & JsSIP expert with VoIP / SIP phones.

Thank you!
Related categories: JavaScript VoIP SIP WebRTC