[OpenSIPS-Users] RTP Delay when changing RTP Source port
Rik Broers
RBroers at motto.nl
Wed Jul 22 13:30:13 CEST 2015
I think rtpproxy_engage doesnt work correct with the fact that you bridge internal to external. Also says in docs:
"... Note that when used in bridge mode, this function might advertise wrong interfaces in SDP (due to the fact that OpenSIPS is not aware of the RTPProxy configuration), so you might face an undefined behavior."
You could try and use the rtpproxy_offer and answer functions. put in the reply route an if (has_body("application/sdp")) to also catch the 183 with sdp .The docs have examples on how to use them and how to trigger on reply routes.
Regards,
Rik
-----Oorspronkelijk bericht-----
Van: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] Namens Trevor Steyn
Verzonden: woensdag 22 juli 2015 9:52
Aan: users at lists.opensips.org
Onderwerp: [OpenSIPS-Users] RTP Delay when changing RTP Source port
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
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