[Spce-user] no voice or video register ok

Daniel Grotti dgrotti at sipwise.com
Fri Dec 29 04:09:45 EST 2017


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
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> https://lists.sipwise.com/listinfo/spce-user




More information about the Spce-user mailing list