[Spce-user] Direct RTP on LAN / Signaling on Internet Host
Alvaro Avello
alvaro.avello at gmail.com
Thu Jul 28 12:20:22 EDT 2016
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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