[OpenSIPS-Users] Confused about rtpproxy_offer() w/s Flags
I'm struggling to understand the w & s flags to rtpproxy_offer etc. The documentation is a little unclear, is there any difference between the two? I seem to have had better results using w but I don't understand why that might be. TIA Mark. -- Mark Farmer farm...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Check config files before stop opensips service
I would like to know, if there is a way to check opensips's config files before stop service. Ex: If I do a syntax error into cfg files (with opensips started), I want to "ban" the daemon shutdown (service opensips stop) I'm playing with opensips.service and opensips -C -f $config to try to reach my goal, but I'm not be able to do at this moment... Do you have an idea please ? Thanks in advance ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy No Audio on Outbound Calls
Mark,
You can detect if the INVITE came from your Asterisk by testing the $si
pseudo-variable.
That will allow you to identify the direction of the call. I usually set a
flag for this purpose. For example:
If ($si == "my.ast.er.isk")
setflag(DIR_OUT);
At the point where you engage the rtpproxy, you will then be able to reverse
the internal/external parameters for the function call depending on the
direction of the call
If (isflagset(DIR_OUT)) {
rtpproxy_offer("corfei");
} else {
rtpproxy_offer("corfie");
}
The same flag should still be valid in the onreply handler where you can do
something similar. [Not sure if I have ie/ei the right way round in my
example].
That said, I'm not sure this topology is a good one to be using.
I would generally try to avoid having the media proxy behind NAT and also
using it in bridging mode - it makes life too complicated.
P.S. Looks like you sorted out the problems with the call to do_routing().
John Quick
Smartvox Limited
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[OpenSIPS-Users] RTPProxy No Audio on Outbound Calls
Hello everyone, all help gratefully received, I've been slogging away at
this for ages!
I have OpenSIPS 2.4.4 & RTPProxy behind 1:1 NAT's (different hosts).
RTPProxy runs so:
/usr/local/bin/rtpproxy -s unix:/var/run/rtpproxy/rtpproxy.sock -u rtpproxy
rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp:10.96.16.58 7722 -l
10.96.0.58 10.98.0.58 -A ext.ip.addr.ess 10.98.0.58 -d DBUG LOG_LOCAL0 -m
1 -M 2
OpenSIPS is sitting between my provider & an Asterisk server which has
phones registered.
When I make calls 'Provider -> OpenSIPS/RTPProxy -> Asterisk -> Phone' all
is good, 2 way audio.
But when the call flows in the opposite direction, I get no audio since SDP
is the same as the 1st call.
How do I get it to reverse the rtpproxy_offer/answer flags?
These are the bits that handles it all:
route[RTPPROXY] {
if (is_method("BYE|CANCEL")) {
rtpproxy_unforce();
}
if (is_method("INVITE")) {
rtpproxy_offer("corwfie");
}
}
onreply_route[DROUTING] {
if (is_method("BYE|CANCEL")) {
sip_trace("tid","d");
rtpproxy_unforce();
}
if ($rs=~"(2[0-9][0-9])") {
rtpproxy_answer("corwfei");
}
}
--
Mark Farmer
farm...@gmail.com
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[OpenSIPS-Users] rtpengine_manage() fails when called from failure_route[] with additional flags.
Hi,
I use only rtpengine_manage() function of
rtpengine_{offer,answer,delete} and it is called from different
locations like request_route, onreply_route, failure_route.
To have everything in one place I call a route[RTPENGINE_MANAGE] which
in its turn prepares rtpengine parameters string (ICE, profiles, flags)
and calls rtpengine_manage(). When this route is called from
failure_route rtpengine_manage() is supposed to behave like
rtpengine_delete() and it does. The only problem is that when it
receives flags in its parameters string (no-rtcp-attribute in my case)
it fails with "rtpengine:parse_flags: error processing flag
`no-rtcp-attribute': no more memory" message instead of just ignoring
useless for delete operation parameters.
Attaching the patch that fixed this problem for me. Not sure if this is
a bug or lack of module documentation.
diff --git a/modules/rtpengine/rtpengine.c b/modules/rtpengine/rtpengine.c
index 2d1a1d3..d9dc4df 100644
--- a/modules/rtpengine/rtpengine.c
+++ b/modules/rtpengine/rtpengine.c
@@ -1666,6 +1666,7 @@ static int parse_flags(struct ng_flags_parse *ng_flags, struct sip_msg *msg,
if (!val.s) {
bitem = bencode_str(bencode_item_buffer(ng_flags->flags), &key);
if (!bitem) {
+LM_ERR("XXX ng_flags->glags => %p\n", ng_flags->flags);
err = "no more memory";
goto error;
}
@@ -1766,8 +1767,10 @@ static bencode_item_t *rtpe_function_call(bencode_buffer_t *bencbuf, struct sip_
ng_flags.to = (op == OP_DELETE) ? 0 : 1;
- if (parse_flags(&ng_flags, msg, &op, flags_str))
- goto error;
+ if (op != OP_DELETE) {
+ if (parse_flags(&ng_flags, msg, &op, flags_str))
+ goto error;
+ }
/* only add those if any flags were given at all */
if (ng_flags.direction && ng_flags.direction->child)
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Re: [OpenSIPS-Users] tls -> udp
Hi Johan,
I've configured Proxy servers to do this a few times.
You should do the following:
1) For Requests going from TLS to UDP, change any occurrence of
"transport=tls" in the R-URI parameters. I use the following to do this:
subst_uri('/transport=tls/transport=udp/I');
2) Make sure OpenSIPS adds correct Record-Route headers.
Default behaviour in this case is to add 2 RR headers when you call
record_route(). Make sure double_rr has not been disabled in the modparam
section.
One header describes the TLS socket and the other header describes the UDP
socket. These are needed for sequential Loose-Routed requests later in the
dialogue.
3) Just before you relay the request over UDP, call the force_send_socket()
function. For example:
force_send_socket(udp:12.34.56.78:5060);
Hope this helps.
John Quick
Smartvox Limited
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