[Freeswitch-users] Inviting both normal and webrtc users

[Jurijs Ivolga]

Hi,

Sorry I bit misunderstood your question, so you can ignore my first reply.
:)

In my implementation I have similar problem when some clients are
connecting via WebRTC and some via SIP.

I solved it in a way that when calls needs to go to WebRTC, then call are
routed to dialplan what is dedicated to WebRTC and plain SIP are routed to
regular dialplan.

This do not answer your question, but I hope it will help. :)

Jurijs


On Wed, Nov 28, 2018 at 2:21 PM Allan Kristensen <ak at hejdu.dk> wrote:

> Hello,
>
> I'm trying to implement calling Webrtc clients (through kamailio proxy),
> which needs ICE, DTLS, etc. in SDP and so we are adding "media_webrtc=true"
> when calling and it works fine.
> But once we do this, many of our other sip clients complain because they
> only support/accept insecure RTP (RTP/AVP).
>
> So question is, shouldn't it be possible to offer both secure and insecure
> RTP in the INVITE from FS?
>
> If not then, would it advisable/possible to not send SDP in the initial
> invite from FS (we don't need early media for inbound anyway) and then wait
> for the client to offer the SDP instead ?
> (I'm afraid the ice process might cause a little silence interval at the
> start of the call because of the late media setup and user would always
> expect media to be up when answering the call)
>
> We really want to have generic way of handling of clients if possible, to
> avoid client detection (as currently only kamailio has this info, doing the
> location services) or any special config items on sip accounts.
>
> Best regards,
>   Allan
>
>
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