[Freeswitch-users] issue of freeswitch working with doubango webrtc clients

Vincent Xia gmangudai at gmail.com
Wed Dec 18 05:05:28 MSK 2013


codec mismatch is not the problem since i had tried PCMA or OPUS at both
ends and fs, now i found this in the webrtc2sip technical guide:

"For example, FreeSWITCH do not support ICE which means it requires the
RTCWeb Breaker inorder to be able to connect the browser to a SIP-legacy
endpoint."
(in the webrtc2sip webpage it's "For example, if your server doesn't
support ICE...", i remember Anthony reminded Mamadou early this year that
FreeSWITCH supports ICE so the webpage is updated)

so i guess the webrtc breaker is not architecturally nessesary and the
issus may be, say, an incompatibility between fs and doubango clients?



2013/12/18 Vincent Xia <gmangudai at gmail.com>

> im testing the interoperability of freeswitch and doubango sip clients
> including boghe and imsdroid, that both have the setting of media profile:
> default or webrtc, setting to default the call is fine as from or to the
> boghe/imdroid client, but setting to webrtc, these clients could only make
> outgoing calls, when receiving calls they respond the fs invite message
> with 488 "bad content", the fs console says the call failed due to
> incompitalbe destination:
>
> 2013-12-17 18:03:50.158267 [NOTICE] switch_ivr_originate.c:2699 Cannot
> create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
> 2013-12-17 18:03:50.158267 [INFO] mod_dptools.c:3201 Originate Failed.
>  Cause: INCOMPATIBLE_DESTINATION
>
> the instruction document from doubango says for the webrtc clients to work
> with the "legacy" sip network a webrtc2sip module is required to as a sip
> proxy and webrtc breaker, but now freeswitch supports webrtc and sip over
> websocket, is a webrtc breaker still mandatory?
>
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