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

[Murugan Pandian]

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
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