[asterisk-users] peer-to-peer RTP trouble in SIP

[Olle E Johansson]

23 feb 2007 kl. 09.52 skrev Michiel van Baak:

> Hey,
>
> We have asterisk 1.2.4 (old I know) with a couple of snom
> phones, a couple of grandstream phones and around 65 philips
> dect stations.
> Now the problem:
> All calls do peer to peer RTP except the calls from dect
> station to dect station.
> snom to dect or dect to snom do peer to peer.
> So the sip config looks fine because all the 'static
> deskphones' honor the REINVITE and start talking to
> eachother.
> Our supplier told us they dont send SDP with the INVITE. Can
> this be the problem causing dect to dect calls to always use
> asterisk in the RTP path ?

If they do not send SDP with the INVITE there will be no
media at all in the call. Very simple.

Can be that they do not support re-invites. If so, you should
see an error message in the SIP communication.

/O