[Freeswitch-users] WebSocket for proxied outgoing calls

[mayamatakeshi]

On Thu, Aug 1, 2013 at 4:32 PM, mayamatakeshi <mayamatakeshi at 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 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?
>

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
>
>
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