Re: [SR-Users] DTLS not working with kamailio
Hello, On 09/04/15 07:17, renuka ghate wrote: I have configured the kamailio server for TLS.. but my soft phone supports only DTLS, how do i configure kamailio to support DTLS. what is the softphone with dtls support? I haven't seen any so for doing sip over dtls. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] DTLS not working with kamailio
I have configured the kamailio server for TLS.. but my soft phone supports only DTLS, how do i configure kamailio to support DTLS. regards, Renuka ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] missing BYE when 2 redundant kamailio servers share the same database
Hi, all
It took some time to clean up the pjsua settings and collect the pcaps for
both cases.
Here are the two pcap files.
call_failed_bye.pcap has the call trace when the BYE is handled with the
following logic.
route[WITHINDLG] {
if (has_totag()) {
if (loose_route()) {
route(DLGURI);
if (is_method(BYE)) {
xlog(L_DBG, =BYE $ru from $fu $si:$sp to $du=\n);
dlg_manage();
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
}
route(RELAY);
This generated a 500 error.
call_extra_bye.pcap is the call trace when the BYE is handled like this:
route[WITHINDLG] {
if (has_totag()) {
if (loose_route()) {
route(DLGURI);
if (is_method(BYE)) {
xlog(L_DBG, =BYE $ru from $fu $si:$sp to $du=\n);
dlg_manage();
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
# If BYE is coming from Kamailio peer, route BYE by location
$var(peerlist)=$sel(cfg_get.trusted.peers);
$var(i) = 0;
while($var(i)$(var(peerlist){param.count})) {
xlog(L_DBG,
=$(var(peerlist){param.count})=$(var(peerlist){param.valueat,$var(i)})=\n);
if(src_ip==$(var(peerlist){param.valueat,$var(i)})) {
lookup(location);
xlog(L_DBG, =BYE from $fu $si:$sp to $du=\n);
break;
}
$var(i) = $var(i) + 1;
}
}
...
route(RELAY);
This got the BYE and 200 OK for the BYE. But there was an extra BYE at the
end, which resulted in 481 Call/Transaction Does Not Exist.
Wonder if the 481 was caused by pjsua not sending ACK for the 200 OK of BYE.
Just to recap the use case. We're trying to set up two kamailio servers
sharing the same database(db_mode=3).
kamailio server 1(k1): 10.0.1.30:5060
kamailio server 2(k2): 10.0.1.32:5060
sip client c1: sip:16317@10.0.1.30, client ip: 10.0.1.254
sip client c2: sip:72316@10.0.1.30, client ip: 10.0.1.254
c1 is registered with k1.
c2 is registered with k2. The expected call flow is as follows:
When c1 calls c2, the expected call flow is as follows:
INVITE:
c1--k1--k2--c2
200 OK:
c1--k1--k2--c2
BYE
c1--k1--k2--c2
But we have the above-mentioned issue with BYE handling at k1.
Thanks
Ding
On Fri, Mar 20, 2015 at 1:08 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
An alternative would be to test with tcp, from routing point of view
should be the same if it is udp-tcp or udp-tls.
Cheers,
Daniel
On 20/03/15 15:09, Vitaliy Aleksandrov wrote:
We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to
get a pcap anyway. Wonder if there is a way to dump pcap from inside
kamailio.
Wireshark can decrypt SIP signalling sent over TLS connections if you
provide server's private key to it.
All the requests within dialog are routed through 2 kamailio instances. We
want to make sure each phone only sends requests through its registrar.
I have included pjsua logs in subsequent emails in this thread. Those
logs have SIP messages, but only provide client perspective.
Thanks for the help,
Ding
On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
On 19/03/15 02:54, Ding Ma wrote:
[...]
My first question is why k1 loose_route sends the BYE to itself instead of
the client. Is this a bug?
can you get the pcap of such call? We have to see the routing headers to
say what is next hop address.
Are all the requests within dialog routed via same instance of kamailio?
My next question is whether the above location routing for BYE from peer
kamailio a good/safe approach.
The SIP traces will be sent later to avoid exceeding email size limit.
Cheers,
Daniel
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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call_failed_bye.pcap
Description: Binary data
call_extra_bye.pcap
Description: Binary data
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[SR-Users] Avoiding script writer didn't release transaction warnings
Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] DTLS not working with kamailio
I think he's referring to media encryption using DTLS-srtp. Like with sipml5 or jssip. I guess a software name and version would help to clear up the question. Nir S On Apr 9, 2015 11:25 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 09/04/15 07:17, renuka ghate wrote: I have configured the kamailio server for TLS.. but my soft phone supports only DTLS, how do i configure kamailio to support DTLS. what is the softphone with dtls support? I haven't seen any so for doing sip over dtls. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Stupid trace and log question
Hello, We are running kamailio 4.1.1 for our services and we noticed a strange issue with logs when we launch the command: /usr/sbin/kamctl fifo debug 2 only INFO level traces appear in the log file With /usr/sbin/kamctl fifo debug 3 only DEBUG level traces appears. How do we activate traces so INFO and DEBUG traces appear ? Emmanuel BUU http://www.ives.fr/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] The caller doesn't receive reject message when the callee has more one register location
Hello, depending on how you have tm configure, the behaviour for parallel forking can be to wait for all branches to complete. If callee has many contacts, then many branches are created and kamailio waits until each branch receives a final response or times out. Have you waited until the other branches timed-out? Can you share all the parameters you set for tm module? Cheers, Daniel On 09/04/15 04:45, xuefeng zhang wrote: Dear Sir/Madame, I find out a caller doesn't receive reject message when a callee has been registered more times with different device then keep only one callee's device is available.I use the linphone's app to test it,It always reproduce on 4.2.4 stable version.The linphone's app reject a new call that will send 603 message,but the caller can't receive the 603 message.The caller will be continue at calling state. So I add a source code *should_relay=branch at 1384 line number in t_reply.c file. Hope you can change it. Thanks! Xuefeng Zhang ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Avoiding script writer didn't release transaction warnings
I think understand in which case it happens. When I receive a REGISTER, I create a transaction since I need to use the t_continue feature to wake up a suspended transaction. However, the save() command of the REGISTRAR module doesn't seem to call t_reply for replying the 200 (it looks like the SL module is used instead). Therefore, I get the warning every time it happens since the transaction is not released explicitly. Should I call t_release just before exiting the block handling the REGISTER? From: Mickael Marrache [mailto:mickaelmarra...@gmail.com] Sent: Thursday, April 09, 2015 1:43 PM To: sr-users@lists.sip-router.org Subject: Avoiding script writer didn't release transaction warnings Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Work with radius
Hello, On 09/04/15 05:36, Никитенко Виталий wrote: Hi. I have scheme: User - Kamailio - Asterisk - PSTN | V Radius Server At the beginning dialog (INVITE) kamailio send to radius server START packet which indicates Called, Calling, Accounting Session ID, connect_time, setup_time. When the terminating call (BYE) kamailio forms STOP package. How can I get on kamailio Called, Calling, connect_time that I have in kamailio for START packet if I only match Accounting Session ID? a short description for sake of completion: acc module for Kamailio is sending START and STOP events, it is up to you to mitigate in the radius backend side and get the call duration. On the other hand, if you want to get Called, Calling, connect_time values from INVITE at BYE time you have to use dialog module and store those values in dialog variables. Then send those values to radius via extra accounting. However, you will get duplicates in START record. If you are not bound to radius, by using acc+dialog modules, you can get one cdr per call in database (or syslog). With a bit of c coding, this should be possible to be extended for RADIUS. Not using RADIUS myself I wasn't able to code it myself. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
