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

[Vincent Xia]

originate user/1001 &echo() works fine with chrome client and the
difference is boght/imdroid has no DTLS HANDSHAKE after call answered:

[INFO] switch_rtp.c:2449 Changing audio DTLS state from HANDSHAKE to SETUP

[INFO] switch_rtp.c:2368 audio Fingerprint Verified.

[INFO] switch_rtp.c:2868 Activating Audio Secure RTP SEND

[INFO] switch_rtp.c:2846 Activating Audio Secure RTP RECV

[DEBUG] switch_core_sqldb.c:2357 Secure Type:
srtp:dtls:AES_CM_128_HMAC_SHA1_80
[INFO] switch_rtp.c:2408 Changing audio DTLS state from SETUP to READY

[DEBUG] switch_core_sqldb.c:2357 Secure Type:
srtp:dtls:AES_CM_128_HMAC_SHA1_80



2013/12/19 Vincent Xia <gmangudai at gmail.com>

> 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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>
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