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