Re: [OpenSIPS-Users] SIP session termination doesn't work properly
Hi Bogdan, I just captured the two sessions with ngrep (attached to this mail). Suprisingly, the dialog between the two laptops is all good with ngrep. But when I use the two mayah centauri hardware codecs, there is still the problem with the "404 - not here". Which brings me back to my assumption that it is not a opensips problem. Can you confirm that? Thanks & best regards, Simon Original-Nachricht > Datum: Mon, 18 May 2009 17:45:54 +0300 > Von: Bogdan-Andrei Iancu > An: simon.wi...@gmx.de > CC: users@lists.opensips.org > Betreff: Re: [OpenSIPS-Users] SIP session termination doesn\'t work properly > Hi Simon, > > could you post the SIP capture with full message (use ngrep on the > machine) - otherwise it it rather impossible to figure out what is going > on there. > > Regards, > Bogdan > > Simon Witte wrote: > > Hi all, > > > > > > I'm having trouble with the proper termination of SIP sessions going > > through my proxy server (opensips 1.5.1) > > There are 2 scenarios so far: > > > > 1) Session between two hardware audiocodec (Mayah Centauri) > > The BYE from one client isn't answered by a "200 - OK" but a "404 - > > Not here". First one client tries 12 times, then the other one. > > > > screenshots from wireshark: > > http://www.vivid-vision.de/Files/bye_404_no1.jpg > > > > http://www.vivid-vision.de/Files/bye_404_no2.jpg > > > > 2) Session between two laptops using a sip-software > > the BYE seems to hop from one client to the other until it times > out. > > > > screenshot from wireshark: > > http://www.vivid-vision.de/Files/bye_timeout.jpg > > > > > > Can somebody please tell if there's something wrong with the routing > > logic or has a hint to what the problem might be? > > I'm not so sure yet, if it's an opensips-problem, but I want to rule it > > out first. > > > > Thank you all in advance. > > > > > > Best regards, > > Simon > > > > > > -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss für nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a interface: eth0 (192.168.1.0/255.255.255.0) filter: (ip) and ( port 5060 ) U 192.168.1.70:5060 -> 192.168.1.60:5060 BYE sip:may...@192.168.1.60 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.70:5060;branch=z9hG4bKWDX3mg5yEJs;rport. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Max-Forwards: 70. Contact: ;expires=1800. Expires: 1800. User-Agent: MAYAH Communications V2.0.0.44. Content-Length: 0. . U 192.168.1.60:5060 -> 192.168.1.70:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 192.168.1.70:5060;branch=z9hG4bKWDX3mg5yEJs;rport=5060. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Server: OpenSIPS (1.5.1-notls (i386/linux)). Content-Length: 0. . U 192.168.1.70:5060 -> 192.168.1.60:5060 ACK sip:may...@192.168.1.60 SIP/2.0. Via: // :342764;branch=z9hG4bKWDX3mg5yEJs;ttl=16726422;rport. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Max-Forwards: 70. User-Agent: MAYAH Communications V2.0.0.44. Content-Length: 0. . U 192.168.1.70:5060 -> 192.168.1.60:5060 BYE sip:may...@192.168.1.60 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.70:5060;branch=z9hG4bKTPAZB1jE96u;rport. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Max-Forwards: 70. Contact: ;expires=1800. Expires: 1800. User-Agent: MAYAH Communications V2.0.0.44. Content-Length: 0. . U 192.168.1.60:5060 -> 192.168.1.70:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 192.168.1.70:5060;branch=z9hG4bKTPAZB1jE96u;rport=5060. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Server: OpenSIPS (1.5.1-notls (i386/linux)). Content-Length: 0. . U 192.168.1.70:5060 -> 192.168.1.60:5060 ACK sip:may...@192.168.1.60 SIP/2.0. Via: // :342764;branch=z9hG4bKTPAZB1jE96u;ttl=16726422;rport. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Max-Forwards: 70. User-Agent: MAYAH Communications V2.0.0.44. Content-Length: 0. . U 192.168.1.70:5060 -> 192.168.1.60:5060 BYE sip:may...@192.168.1.60 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.70:5060;branch=z9hG4bKxxvgwh6HsCY;rport. To: ;tag=xQluw0HeNt4. From: ;tag=GcgcQtcWLQi. Call-ID: xqluw0he1242662...@192.168.1.50. CSeq: 2 BYE. Max-Forwards: 70. Contact: ;expires=1800. Expires: 1800. User-Agent: MAYAH Communications V2.0.0.44. Content-Length: 0. . U 192.168.1
[OpenSIPS-Users] SIP session termination doesn't work properly
# 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
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authenticate if from local subscriber (uncomment to enable auth)
# authenticate all initial non-REGISTER request that pretend to be
# generated by local subscriber (domain from FROM URI is local)
if (!(method=="REGISTER") && from_uri==myself) /*no multidomain
version*/
##if (!(method=="REGISTER") && is_from_local()) /*multidomain version*/
{
if (!proxy_authorize("192.168.1.60", "subscriber")) {
proxy_challenge("192.168.1.60", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
}
# 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
if (!is_method("REGISTER"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
##t_relay("tls:domain1.net");
##exit;
##} else if($rd=="tls_domain2.net") {
##t_relay("tls:domain2.net");
##exit;
##}
route(1);
}
# requests for my domain
## uncomment this if you want to enable presence server
## and comment the next 'if' block
## NOTE: uncomment also the definition of route[2] from below
##if( is_method("PUBLISH|SUBSCRIBE"))
##route(2);
if (is_method("PUBLISH"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("192.168.1.60", "subscriber"))
{
www_challenge("192.168.1.60", "0");
exit;
}
if (!check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence
NOTE: do not forget to enable the call of this route from the main
route */
##route[2]
##{
##if (!t_newtran())
##{
##sl_reply_error();
##exit;
##};
##
##if(is_method("PUBLISH"))
##{
##handle_publish();
##t_release();
##}
##else
##if( is_method("SUBSCRIBE"))
##{
##handle_subscribe();
##t_release();
##}
##
##exit;
##}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404","Not found");
##exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status("486|408")) {
##sethostport("192.168.2.100:5060");
### do not set the missed call flag again
##t_relay();
##}
}
--
___
Simon Witte
phone: +49 211 - 545 888 02
mobil: +49 172 - 286 20 76
mail: simon.wi...@gmx.de
Engelbertstr. 11 | 40233 Düsseldorf
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[OpenSIPS-Users] opensips 1.5.1 - how to update the MySQL-DB if IP-address of UA changes
Dear sirs and madams, I'm a student from germany and currently I'm trying to install a SIP-server. This is my first time working with SIP (in general) and opensips. Hopefully I won't embarass myself too much. So here's my problme: Is there a way to implement a routine to update the MySQL-DB faster if the IP-address of a client changes? To be more specific: In the first step of my project I'm using only 192.168.1.x as my domain. The Proxy/ Registar can be found at 192.168.1.60 and I'm using two PCs as UACs. I can register a client with the server, if he's at e.g. 192.168.1.200. If I change its IP-address (to e.g .180), the changes take quite some time to manifest. If I re-register manually (ekiga sopftphone), there are two entries in the location-table, which I certainly don't want. On the one hand, how do I prohibid a UAC from registering with two different IP-addresses using the same username/ password. And on the other hand how can I speed up the process to update the database when a IP-address change takes place?! I'd really appreciate your help and thank you in advance. Best regards, Simon Witte -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss für nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
