[OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri

juancarlosg6 juancarlosg6 at gmail.com
Wed Feb 3 21:48:26 EST 2021

Hi Mark,  is working for us more or less, because is strange situtation, if
the extension WSS received a incoming call working good but if a put the
call in on-hold inmediatly hangout, but if the same extension do the call to
a external number or a SIP extension or another Webrtc extension work very
good hold and unhold very good the call, i dont know how to identify what is
the problem, we are trying to do this in Opensips because actually our
Asterisk PBX receiving directly WSS conections some times have Flooding
Error (From Traffic Valid).

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