On Thu, Aug 1, 2013 at 4:32 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
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@test1.com>;tag=6197r4g0Kar9H. To: <sip:user2@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?
I have found the answer on this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097369.html Set *media_webrtc=true* when originating the call. Sorry for the noise. I will document the var at the wiki. 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