14 Apr
2010
14 Apr
'10
3:27 p.m.
Hi!
1. it is useful to find out if the problem happens due to a T.38
reINVITE or with all reINVITEs. You could easily test this by just
putting a call on-hold, and back off-hold (e.g. by pressing the line
button in Xlite) and verify if audio works again.
more inline ...
Am 14.04.2010 11:28, schrieb Espen Berg:
Hello all!
I finally have something that I could call a working config (almost that
is), the proxy works fine with voice and fax sent over g.711 (but I
guess re-invites will cause problems in this scenario also). The
problem occurs when I try to switch to T.38. Since it works with voice
my guess is that the problem is caused by the re-invite to t.38. I
guess it has to be solved by a if (loose_route()) section, but I am
little bit clueless right now, I have tried multiple variants but with
no success.
RTP proxy is started with the -l flag to distinguish between local and
public traffic.
-l PUBLIC.IP.ETH0/LOCAL.IP.ETH1
I would reverse the interfaces as the first interface corresponds with
the "internal" interface and the second with the "external". See
'ie'
flags at
http://sip-router.org/docbook/sip-router/branch/master/modules_k/nathelper/…
When a call is routed from local IP to public IP I would like to name it
internal-to-external, hence 'ie' flags instead of 'ei', but that s just
a matter of naming and shouldn't cause problems.
The setup:
Sonus UAC/Trunk <-private ip-> (eth1)Kamailio/RTPProxy(eth0) <-public
ip-> Asterisk
IP.TO.TRUNK = Sonus IP, public address range, but only reachable by
private net (from the range of LOCAL.IP.ETH1).
LOCAL.IP.ETH1 = Local net, static route to IP.TO.TRUNK.
PUBLIC.IP.ETH0 = Public routable address.
IP.TO.ASTERISK = Public routable address.
The purpose of Kamailio/RTPProxy are to proxy everything between the
routable and the private net.
Does anyone here have any idea of how I'm going to solve my issue? It
would also be nice if anyone here have better solutions to my approach,
and there could also be some other errors in my config that causes this
behaviour.
####### Routing ########
route {
if (msg:len >= 2048) {
sl_send_reply("513", "Message too big");
exit;
}
if (method != "INVITE" && method != "ACK" && method
!= "BYE" &&
method != "CANCEL") {
sl_send_reply("503", "Service Unavailable");
exit;
}
if (!has_totag()) {
record_route();
route("1");
}
if (has_totag()) {
if (!loose_route()) {
if (t_check_trans()) {
t_relay();
exit;
}
exit;
}
}
That's strange. You shouldn't allow requests with totag but without
loose-routing. But that's not related to your problem.
if (has_body("application/sdp")) {
if (method == "INVITE") {
route("2");
t_on_reply("2");
} else if (method == "ACK") {
rtpproxy_answer();
}
} else if (method == "INVITE") {
t_on_reply("2");
}
if (method == "BYE" || method == "CANCEL") {
unforce_rtp_proxy();
}
t_relay();
}
#RELAY BY SOURCE ROUTE
route[1] {
if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") {
force_send_socket("PUBLIC.IP.ETH0");
rewritehost("IP.TO.ASTERISK");
} else {
force_send_socket("LOCAL.IP.ETH1");
rewritehost("IP.TO.TRUNK");
}
}
# RTP OFFER ROUTE
route[2] {
if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1")
force_rtp_proxy("eir");
else
force_rtp_proxy("ier");
}
# RTP OFFER ONREPLY
onreply_route[1] {
if (has_body("application/sdp"))
route("2");
}
# RTP ANSWER ONREPLY
onreply_route[2] {
if (has_body("application/sdp"))
route("2");
}
## END OF CONFIG
Sorry for the long post, but I hope someone outthere are able to help
med with this issue.
IMO the config is a bit strange but I couldn't spot an error.
Trace the scenario with ngrep and take a look at the SDPs if they are
rewritten properly.
regards
klaus