Use “rtp set debug on” at the CLI and see in which direction and on which legs (four in total) the RTP is missing.
Also, please use logging obtained from Asterisk itself (verbose level 5 and the appropriate protocol logging option for your channel driver. Please say which channel driver that is.
The channel driver configuration is also likely to be needed, as most such problems are the result of errors in that.
Unfortunately, WebRTC and therefore ICE go beyond my experience, so I may nto be able to provide much help on the RTP port negotiation.
The SIP Channel Driver setting is chan_pjsip and the configuration :
pjsip.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------; #include pjsip_custom.conf #include pjsip.transports.conf #include pjsip.transports_custom_post.conf #include pjsip.endpoint.conf #include pjsip.endpoint_custom_post.conf #include pjsip.aor.conf #include pjsip.aor_custom_post.conf #include pjsip.auth.conf #include pjsip.auth_custom_post.conf #include pjsip.registration.conf #include pjsip.registration_custom_post.conf #include pjsip.identify.conf #include pjsip.identify_custom_post.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------; #include pjsip.transports_custom.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------; #include pjsip.endpoint_custom.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------; #include pjsip.aor_custom.conf
I changed the docker image from ’ tiredofit/freepbx’ to ’ izdock/izpbx-asterisk’
It worked fine on WebRTC client(Desktop) to SIP Client(Linphone-Android)
but not on WebRTC client(Desktop) to WebRTC client(Mobile) https://pastebin.freepbx.org/view/73f7987a