[Freeswitch-users] Early Media,Hangup issue in WebRTC

14 Oct 2016

      Hi,

  I am using SIPJS for my webrtc client ,When i try to make call pstn, i
 cant able to hear any early  media and if b leg (pstn) disconnect call
,A-leg(sipjs) did't receive bye message.

if i use this same scenario  in my voipfone(xlite.zoiper) works fine.

Regard's
Murugan Pandian