[SR-Users] Rtpengine - no RTP packets going out

[Istvan Mogyorosi]

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