This is how I configured rtpproxy. Not sure why rtpproxy is not engaged both ways. Thanks for your help.
root@Kamailio:/etc/default# more rtpproxy # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: LISTEN_ADDR=public ip address CONTROL_SOCK="udp:localhost:22222" # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" root@Kamailio:/etc/default# ##kamailio.cfg## #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pin...@mydomain.com") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } On Fri, Jun 3, 2011 at 4:33 AM, Alex Balashov <abalas...@evaristesys.com> wrote: > This sounds like an asymmetrical engagement of rtpproxy (e.g. on the reply > but not the initial request, or vice versa). > > On 06/02/2011 09:45 PM, Mokhtar Bengana wrote: > >> I have Kamailio and rtpproxy running behind NAT on the same server. >> Inbound calls to Kamailio from the outside work fine and I have two >> way audio but when I try an outbound echo test call the call connects >> but I have no audio. rtpproxy seems to be running and all rtp ports >> are poiting to Kamailio internal ip. The syslog is showing this error. >> Not sure what's missing in the config. Appreciate any inputs. Thanks. >> >> >> >> root@Kamailio:~# tail /var/log/syslog >> Jun 3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy >> [rtpproxy.c:1403]: rtp proxy<udp:127.0.0.1:22222> found, support for >> it enabled >> Jun 3 01:25:05 Kamailio kamailio[791]: INFO:<core> [forward.c:786]: >> broken reply to forward - no 2nd via >> Jun 3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy >> [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy >> Jun 3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy >> [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users