[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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