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

[Vincent Xia]

some updates:

i found the problem is that boghe/imsdroid sends sip (including REGISTER)
over udp, thus taken by fs as a "legacy" endpoint, so the INVITE sent to
them contains a "legacy" style sdp whereas they require a "webrtc" style
one. i think that's why 488 is generated.

normally the webrtc application takes place between web browsers using sip
over websocket, the "legacy" endpoints using sip over udp, but
boghe/imsdroid are special, they use sip over udp and supports webrtc, so
my question is:

is it possible to have fs send "webrtc" style sdp to
sip-over-udp-endpoints? any clue?



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

> 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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