[Freeswitch-users] WebSocket for proxied outgoing calls

[mayamatakeshi]

Hello,
I have kamailio SIP proxy with WebSocket support in front of FS using plain
UDP transport (not using ws-binding).
It works fine for incoming calls:
http://sipml5.org --> SIP over WebSocket --> kamailio --> SIP over UDP -->
FS plays some prompt.

Then, the WebSocket app registers with kamailio and when a call arrives to
it at FS, FS sends the call to kamailio and the call is sent to the browser.
However, when I try to answer the call, the web app refuses the call with
this:

SIP/2.0 603 Failed to get local SDP.
From: "displayname.user3"<sip:user3 at test1.com>;tag=6197r4g0Kar9H.
To: <sip:user2 at 127.0.0.1>;tag=okDK7cd5n34HgaEM9dgs.
Call-ID: 2d1ab230-7516-1231-969f-5254000fd208.
CSeq: 47333138 INVITE.
Content-Length: 0.

I suppose this is happening because an SDP for a WebRTC call cannot be a
plain one that we send on ordinary calls. Is this correct?
So is it possible to instruct FS (some parameter when calling originate or
bridge) to prepare an SDP for WebRTC?
If not, I was thinking if asking FS to not send SDP in the INVITE would
work (late negotiation) as it would receive the SDP in the "200 OK" and
send its SDP in the ACK. But i could not find a way to force FS to do this
either.

regards,
Takeshi
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