Dear all,
This is my first post after reading a lot in this mailing-list.
I'm trying to use Kamailio 5.1 with the dispatcher module and rtpengine
acting as SIP + RTP proxy.
I have 6 asterisk servers in a private subnet that should talk with the
peer via a single IP like this:
Asterisk 1..n|---> | GW.PRIVATE.IP -o- GW.PUBLIC.IP |----> PEER.SIP.TRUNK
I'm on Centos 7, with firewalld configured, iptables module is loaded
and the rule is well defined.
Packet forwarding is also enabled.
Chain rtpengine (1 references)
target prot opt source destination
RTPENGINE udp -- anywhere anywhere RTPENGINE
id:40
My call flow seems to be fine, Kamailio/rtpengine private IP is the
outboundproxy parameter of Asterisk instances.
My problem is that RTP packets are not present on the public interface,
the rtpengine final log showing
the 2 sessions, but I'm not sure this is what I want or simply the
firewall does not let it out ?
(To be more precise PEER.SIP.TRUNK is the trunk for SIP traffic, I have
multiple IP addresses
for media to connect to, reinvites are allowed)
Closing call due to timeout
Final packet stats:
--- Tag 'as6d12caea', created 1:00 ago for branch '', in dialogue with
'as541b1e61'
------ Media #1 (audio over RTP/AVP) using unknown codec
--------- Port GW.PRIVATE.IP:10000 <> 192.168.30.13:11152, SSRC 0, 0
p, 0 b, 0 e, 60 ts
--------- Port GW.PRIVATE.IP:10001 <> 192.168.30.13:11153 (RTCP),
SSRC 0, 0 p, 0 b, 0 e, 60 ts
--- Tag 'as541b1e61', created 1:00 ago for branch '', in dialogue with
'as6d12caea'
------ Media #1 (audio over RTP/AVP) using unknown codec
--------- Port GW.PUBLIC.IP:10000 <> PEER.SIP.TRUNK:28216, SSRC 0,
0 p, 0 b, 0 e, 60 ts
--------- Port GW.PUBLIC.IP:10001 <> PEER.SIP.TRUNK:28217 (RTCP),
SSRC 0, 0 p, 0 b, 0 e, 60 ts
Best regards,
Istvan
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