Hi, You should check SDP in SIP Invites to see where it sends audio. Maybe it points to a wrong IP/port It's easy to do with sngrep / asterisk pjsip history
Hope it helps
-----Mensaje original----- De: fernandolopes20003--- via sr-users sr-users@lists.kamailio.org Enviado el: lunes, 7 de abril de 2025 16:48 Para: sr-users@lists.kamailio.org CC: fernandolopes20003@gmail.com Asunto: [SR-Users] Kamailio + RTP Engine question
Hi everyone,
I'm experiencing some issues with Kamailio and RTP Engine.
Here's the setup: A user initiates a call, and the traffic goes through a router. The router forwards the traffic to a machine at 192.168.1.75, which runs Kamailio and RTP Engine. Kamailio uses a dispatcher to forward calls to 192.168.1.190, a Kubernetes node running an Asterisk container.
The problem: When I'm inside the network, everything works fine — I can send and receive audio. But when I'm outside the network, the call connects, but there's no audio in either direction (no send or receive).
Some Configs: Kamailio.conf: modparam("rtpengine", "rtpengine_sock", "udp:0.0.0.0:2223") listen=udp:192.168.1.75:5060 listen=tcp:192.168.1.75:5060 listen=udp:127.0.0.1:5060 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 60) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_timer_mode", 1)
rtpengine.conf: interface = 192.168.1.75 listen-udp = 0.0.0.0:2223 port-min = 10000 port-max = 20000
Firewall: To Action From -- ------ ---- 22 ALLOW Anywhere 12222/udp ALLOW Anywhere 5060/udp ALLOW Anywhere 10000:20000/udp ALLOW Anywhere 22 (v6) ALLOW Anywhere (v6) 12222/udp (v6) ALLOW Anywhere (v6) 5060/udp (v6) ALLOW Anywhere (v6) 10000:20000/udp (v6) ALLOW Anywhere (v6)
Also router has all this ports open
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