Web rtc calling problem

Hi Team,

I have a problem when using WEB RTC
i can able to make a call from webrtc user to normal sip user, but then disconnected the call soon.Also, i got a warning error from the console ,I attached screen shot please help me

The messages state the problem. One side is ulaw, one side is opus. You have no module installed to transcode between the two, so the call terminates. You can either install codec_opus by selecting it in “make menuselect” or using a different codec than opus.

Thanks for your valubale reply ,let me check and update you

Hi ,

Now webrtc is working ,thanks for your great support.

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.