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