Hello:
I am using portsip webRTC gateway to test the compatibility.
I use two phones for webRTC testing. One is normal webRTC client
and another one is softphone.
FreeSWITCH work as a IPPBX. The
call flows are:
the problem is that webRTC calls sip phone without any
problem, but sip phone calls to webRTC than failed.
errors are:
2016-08-23 16:29:39.971775 [DEBUG]
switch_core_state_machine.c:569
(sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid) State Change
CS_REPORTING -> CS_DESTROY
freeswitch@iZ23lkvsnwpZ>
2016-08-23 16:29:39.971775 [DEBUG]
switch_core_session.c:1647 Session 171
(sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid) Locked, Waiting
on external entities
freeswitch@iZ23lkvsnwpZ>
2016-08-23 16:29:39.981772 [DEBUG]
switch_ivr_originate.c:3759 Originate Resulted in Error Cause:
88 [INCOMPATIBLE_DESTINATION]
freeswitch@iZ23lkvsnwpZ>
2016-08-23 16:29:39.981772 [NOTICE]
switch_ivr_originate.c:2771 Cannot create outgoing channel of
type [user] cause: [INCOMPATIBLE_DESTINATION]
freeswitch@iZ23lkvsnwpZ>
2016-08-23 16:29:39.981772 [DEBUG]
switch_ivr_originate.c:3759 Originate Resulted in Error Cause:
88 [INCOMPATIBLE_DESTINATION]
freeswitch@iZ23lkvsnwpZ>
2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401
Originate Failed. Cause: INCOMPATIBLE_DESTINATION
freeswitch@iZ23lkvsnwpZ>
EXECUTE sofia/internal/1004@120.26.130.142 answer()
-------------------------
It looks freeswitch send with ICE to webRTC. Does anyone know
how i do disable the webRTC negotiation and only send normal sip
info to webRTC gateway
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