Hi, All Still quite new to opensips I have the following configuration running on
version: opensips 2.1.0 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 06:22:03 May 8 2015 with gcc 4.4.7 (Topology Hiding) UAC -------> Opensips(Internal) Opensips(External) ----> UAS (RTP PROXY BRIDGE) what i am experiencing is the following call is setup between UAC and UAS through opensips UAS sets up RTP with a 183 Session Progress message with SDP Shortly after we get a 180 ringing (i understand this is not correct but cannot be changed), When a 200OK is eventually sent the Source Port is different to what was in the SDP on the 183 message. Media starts flowing from UAS to opensips External from the new source port but for the first 7 seconds or so opensips/rtpproxy does not pass on this media to UAC from Internal. I run rtp proxy as follows rtpproxy -l <Internal IP>/<External IP> -s udp:127.0.0.1:12221 -m 25000 -M 65000 -F -d DBUG:LOCAL1 route{ #script_trace( 3, "$rm from $si, ruri=$ru", "me"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if ( check_source_address("1","$avp(trunk_attrs)") ) { # request comes from trunks setflag(IS_TRUNK); } else if ( is_from_gw() ) { # request comes from GWs } else { send_reply("403","Forbidden"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if(topology_hiding_match()) { # validate the sequential request against dialog if ( $DLG_status!=NULL && !validate_dialog() ) { xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n"); ## exit; } if (is_method("BYE")) { setflag(ACC_DO); # do accounting ... setflag(ACC_FAILED); # ... even if the transaction fails } else if (is_method("INVITE")) { # even if in most of the cases is useless, do RR for # re-INVITEs alos, as some buggy clients do change route set # during the dialog. record_route(); } route(RELAY); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction -> # ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } #### INITIAL REQUESTS if ( !isflagset(IS_TRUNK) ) { ## accept new calls only from trunks send_reply("403","Not from trunk"); exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } else if (!is_method("INVITE")) { send_reply("405","Method Not Allowed"); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } t_check_trans(); # preloaded route checking if (loose_route()) { xlog("L_ERR", "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); if (!is_method("ACK")) sl_send_reply("403","Preload Route denied"); exit; } # record routing record_route(); setflag(ACC_DO); # do accounting # create dialog with timeout if ( !create_dialog("B") ) { send_reply("500","Internal Server Error"); exit; } dp_translate("1","$rU/$rU"); # route calls based on prefix if ( !do_routing("1",,,,"$var(gw_attributes)") ) { send_reply("404","No Route found"); exit; } if (is_method("INVITE")) { force_send_socket(udp:<EXternal IP:5060); rtpproxy_engage('ierz20'); #rtpproxy_engage(); topology_hiding(); } t_on_failure("GW_FAILOVER"); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[GW_FAILOVER] { if (t_was_cancelled()) { exit; } # detect failure and redirect to next available GW if (t_check_status("(408)|([56][0-9][0-9])")) { xlog("Failed GW $rd detected \n"); if ( use_next_gw() ) { t_on_failure("GW_FAILOVER"); t_relay(); exit; } send_reply("500","All GW are down"); } } local_route { if (is_method("BYE") && $DLG_dir=="UPSTREAM") { acc_log_request("200 Dialog Timeout"); } } Below you can see the call flow http://salamander.iburst.co.za:8000/personal/signalling.txt I have tried a most of the options on rtpproxy_engage with no luck Regards Trevor Steyn _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users