[Spce-user] no voice or video register ok

[Michael Stockenhuber]

Dear Daniel,
Thank you very much for the tips and sorry for my late reply. I am at the other
side of the world in down under. Sngrep seems like a great tool (thank you very much for the tip). I hope I understand what the output means.
I changed the rtpproxy settings and interface and this was somewhat of an
improvement. I now have most of the time audio from the smartphone (outside
network) to the doorstation ( local network).
I had video a couple of tries but most of the time this is a no.
>From the doorstation (internal ip) to the smartphone the flow ends at
the router and at the external ip of the server.
Cheers
Michael

> On December 29, 2017 at 8:09 PM Daniel Grotti <dgrotti at sipwise.com> wrote:
> 
> 
> Hi Michael,
> it seems an RTP issue. A can call B, B rings and can pickup, but then no 
> audio. Is this the situation?
> 
> Please check the following parameters in your Domain preferences and 
> make sure they are set to the following values:
> 
> 1) use_rtpproxy = Always with plain SDP
> 2) rtp_interface = ext
> 3) transport_protocol = transparent
> 
> If yes, then it could be a network issue, RTP stream cannot flow 
> properly from A <-> SPCE <->B.
> To investigate that, you would need to check a SIP/RTP trace. You can 
> use command "sngrep -r" to make this trace, and see why RTP cannot reach 
> the end parties.
> 
> Daniel
> 
> 
> 
> On 12/29/2017 08:55 AM, michaels wrote:
> > Dear mailing list,
> >
> > I have installed sipwise on a debian machine and all seems to work well (
> > after quite a few hitches). I have installed a domain and users and
> > subscribers. All I want to do is to use a number of sip phones communicating
> > with each other. I can register an entrance door station and a mobile phone
> > to communicate with the server. I can also ring from one to the other, with
> > the phone being connected to either the internet ( via NAT from the service
> > provider) or with a local IP. ( so i would not think there is a NAT issue).
> > However, I fail to be able to communicate with either voice or video. I can
> > ring, I can pick up from either the phone or the doorstation but there is no
> > communication.
> > I do not have a rewrite rule setup, is that necessary? If I understand
> > correctly this is really for the connection to a PBX.
> > I also have no peering rules setup, again is that necessary for this
> > scenario?
> > I am really new to SIP and Iam working on this for quite some time now, so
> > any help would be appreciated.
> > Cheers
> > Michael
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> 
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