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

[Vincent Xia]

media_webrtc does resolve this issue, but now there's no audio at both
sides, i also tried originate user/1001 &echo() at fs console but no luck.


2013/12/19 Michael Jerris <mike at jerris.com>

> https://wiki.freeswitch.org/wiki/Variable_media_webrtc
>
> On Dec 18, 2013, at 2:56 AM, Vincent Xia <gmangudai at gmail.com> wrote:
>
>
> 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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