[Spce-user] Direct RTP on LAN / Signaling on Internet Host
Raúl Alexis Betancor Santana
rabs at dimension-virtual.com
Thu Jul 28 14:02:07 EDT 2016
Putting a sipwise instance on a private IP behind NAT its calling for problems, if you don't know how to correctly setup it.
> De: "Alvaro Avello" <alvaro.avello at gmail.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Jueves, 28 de Julio 2016 17:20:22
> Asunto: [Spce-user] Direct RTP on LAN / Signaling on Internet Host
> Hi guys. We are building an proof of concept for internal use of sipwise and
> have some issues dealing with an architecture.
> Let's say we have an office with a couple of yealink phones and a gateway FXO
> for some lines. On an external datacenter, we publish to internet a sipwise
> instance in order to allow registrations and deal with signaling.
> So far, we are able to register phones, but each time we make a call, we have
> one way audio. if we use sngrep, we see two calls in state "CALL SETUP", but if
> you pick up the call, just one way audio.
> Our experience in asterisk + yealink phones, the setup is fairy simple. You set
> the peers to allow direct rtp, and even if they are behind nat, they can
> register and the rtp flow is normal.
> our network setup :
> /etc/ngcp-config/network.yml
> ---
> hosts:
> self:
> dbnode: '1'
> eth0:
> ip: 10.1.2.21
> netmask: 255.255.255.0
> type:
> - web_ext
> - sip_ext
> - rtp_ext
> - web_int
> - ssh_ext
> interfaces:
> - lo
> - eth0
> lo:
> cluster_sets:
> - default
> ip: 127.0.0.1
> netmask: 255.255.255.0
> shared_ip: []
> shared_v6ip: []
> type:
> - sip_int
> - ha_int
> - aux_ext
> - ssh_ext
> - api_int
> v6ip: '::1'
> role:
> - proxy
> - lb
> - mgmt
> - rtp
> - db
> Attached some screenshots of sngrep and costumer configuration
> Thanks in advance.
> Saludos / Regards,
> Alvaro
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