[SR-Users] Internet RTP filters
Hi everybody, First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us. We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine. This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up). In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams: - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port) - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port) As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine. However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them. Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result. How could we solve this? Thanks in advance and best regards, Ricardo Dominguez Universidad Miguel Hernández de Elche ___ 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] Internet RTP filters
Hi Andreas, I've tried with TLS (using TCP port 5061) with no RTP encryption and with SRTP, but I don't know what minimum encryption means. Regards, Ricardo -Mensaje original- De: sr-users-boun...@lists.sip-router.org en nombre de andreas kaschner Enviado el: jue 30/06/2011 22:04 Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailingList Asunto: Re: [SR-Users] Internet RTP filters hi, have you tried TLS with Minimum encryption? regards Andreas On 30. juni 2011, at 19:43, Dominguez Jover, Ricardo djo...@umh.es wrote: Hi everybody, First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us. We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine. This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up). In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams: - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port) - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port) As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine. However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them. Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result. How could we solve this? Thanks in advance and best regards, Ricardo Dominguez Universidad Miguel Hernández de Elche ___ 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 ___ 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] Kamailio 3.1.3 uncontrolled core dump
Hi everybody, In the config file I had set debug level to 9 (debug=9). If I change it to its default debug=2, then Kamailio doesn´t crash. For example, if I call a non existent domain it crashes. Is there any reason for this behaviour? Regards, Ricardo -Mensaje original- De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Klaus Darilion Enviado el: viernes, 13 de mayo de 2011 14:14 Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List Asunto: Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump 1. seems like your config is broken and request uri is set to sip:@192.168.64.36 2. Anyways, Kamailio shouldn't crash. I tried to reproduce the failure, but my Kamailio does not crash. regards klaus Am 13.05.2011 10:19, schrieb Dominguez Jover, Ricardo: 06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: core [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: @192.168.64.36 ___ 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] Kamailio 3.1.3 uncontrolled core dump
Hi everybody. As posted in http://lists.sip-router.org/pipermail/sr-users/2011-May/068514.html we are having an uncontrolled core dump. It seems that the crash occurs when some users registers to Kamailio. After upgrading to Kamilio release 3.1.3 we are able to generate the following GDB: GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-32.el5_6.2) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/local/kamailio-3.1.3/sbin/kamailio...done. Reading symbols from /lib/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libresolv.so.2 Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /lib/ld-linux.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so Reading symbols from /usr/lib/mysql/libmysqlclient.so.15...(no debugging symbols found)...done. Loaded symbols for /usr/lib/mysql/libmysqlclient.so.15 Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Reading symbols from /lib/libcrypt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypt.so.1 Reading symbols from /lib/libnsl.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /lib/libm.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libm.so.6 Reading symbols from /lib/libssl.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libssl.so.6 Reading symbols from /lib/libcrypto.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypto.so.6 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1 Reading symbols from /usr/lib/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libgssapi_krb5.so.2 Reading symbols from /usr/lib/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libkrb5.so.3 Reading symbols from /lib/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libcom_err.so.2 Reading symbols from /usr/lib/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libk5crypto.so.3 Reading symbols from /usr/lib/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libkrb5support.so.0 Reading symbols from /lib/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libkeyutils.so.1 Reading symbols from /lib/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libselinux.so.1 Reading symbols from /lib/libsepol.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libsepol.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/pv.so...done. Loaded symbols for
Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump
Hi Klaus, I'm investigating why my config tries to solve @192.168.64.36, I know it is related to presence subscriptions. Many @192.168.64.36 petitions are being processed, however Kamailio only crashes when it comes from one of our users. ¡! I'm working on it Anyway I think that Kamailio shouldn't crash because of this sip uri, if you are trying to reproduce it and need any more logs or debug info please tell me. Regards, Ricardo -Mensaje original- De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Klaus Darilion Enviado el: viernes, 13 de mayo de 2011 14:14 Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List Asunto: Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump 1. seems like your config is broken and request uri is set to sip:@192.168.64.36 2. Anyways, Kamailio shouldn't crash. I tried to reproduce the failure, but my Kamailio does not crash. regards klaus Am 13.05.2011 10:19, schrieb Dominguez Jover, Ricardo: 06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: core [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: @192.168.64.36 ___ 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
Re: [SR-Users] core dump
Thanks Klaus, I found the commit which is included in release 3.1.3. We've upgraded some hours ago and no crash at the moment (so no core dump is neither generated). I'll let you know about this issue. Regards, Ricardo -Mensaje original- De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Klaus Darilion Enviado el: miércoles, 11 de mayo de 2011 11:40 Para: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] core dump IIRC there were some changes in recent kernels which needs some more tweaking (there was a commit from Daniel, but I can't remember the details) regards Klaus Am 11.05.2011 10:28, schrieb Dominguez Jover, Ricardo: Hi everybody, We are having an uncontrolled crash in Kamailio 3.1: May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23366]: : core [pass_fd.c:293]: ERROR: receive_fd: EOF on 39 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: core [main.c:741]: child process 23364 exited by a signal 11 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: core [main.c:744]: core was not generated I've read in several posts I must generate the Core Dump to know what is happening. I'm trying to generate it in this way. /etc/init.d/Kamailio file: if test $DUMP_CORE = yes ; then 1. set proper ulimit ulimit -c unlimited directory for the core dump files COREDIR=/dumps/ [ -d $COREDIR ] || mkdir $COREDIR chmod 777 $COREDIR echo $COREDIR/core.%e.sig%s.%p /proc/sys/kernel/core_pattern fi /etc/default/Kamailio file: DUMP_CORE=yes But the core is not yet generated. Then I've added -w option: OPTIONS=-P $PID_FILE -m $MEMORY -u $USER -g $GROUP -w /dumps/ But core still not generated In kamailio.cfg I´ve also added: disable_core_dump=no User running Kamailio is kamailio who has 777 permissions in /dumps/ directory. No way. Anyhelp would be appreciated. Cheers, Ricardo Domínguez ___ 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 ___ 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] core dump
Thanks Hervé. I'm using Kamailio 3.1.0 in a RHEL 5. Do you know if this issue
is solved in Kamailio release 3.1.3?
Cheers,
Ricardo Domínguez
De: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Hervé Cochet
Enviado el: miércoles, 11 de mayo de 2011 11:20
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] core dump
Hi,
I also have the same problem with kamailio 3.1
I made a modification to the file daemonize.c because if lim.rlim_cur is set to
-1 the test with size parameter at line 491 do not work because rlimit
parameters are unsigned int.
--- kamailio-3.1.0/daemonize.c.ori 2011-04-12 12:24:14.0 +0200
+++ kamailio-3.1.0/daemonize.c 2011-04-12 12:24:57.0 +0200
@@ -488,7 +488,7 @@
strerror(errno));
goto error;
}
- if (lim.rlim_cursize){
+ if ((int)lim.rlim_cursize){
/* first try max limits */
newlim.rlim_max=RLIM_INFINITY;
newlim.rlim_cur=newlim.rlim_max;
Thanks to this patch the core dump should be generated.
This work for me on my testing servers, BUT with my production servers (debian
5.0.1) where the core is not generated and I cannot understand why...
Hervé
On 11/05/2011 10:28, Dominguez Jover, Ricardo wrote:
Hi everybody,
We are having an uncontrolled crash in Kamailio 3.1:
May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23366]: : core
[pass_fd.c:293]: ERROR: receive_fd: EOF on 39
May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT:
core [main.c:741]: child process 23364 exited by a signal 11
May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT:
core [main.c:744]: core was not generated
I've read in several posts I must generate the Core Dump to know what is
happening. I'm trying to generate it in this way.
/etc/init.d/Kamailio file:
if test $DUMP_CORE = yes ; then
1. set proper ulimit
ulimit -c unlimited
directory for the core dump files
COREDIR=/dumps/
[ -d $COREDIR ] || mkdir $COREDIR
chmod 777 $COREDIR
echo $COREDIR/core.%e.sig%s.%p /proc/sys/kernel/core_pattern
fi
/etc/default/Kamailio file:
DUMP_CORE=yes
But the core is not yet generated.
Then I've added -w option:
OPTIONS=-P $PID_FILE -m $MEMORY -u $USER -g $GROUP -w /dumps/
But core still not generated
In kamailio.cfg I´ve also added:
disable_core_dump=no
User running Kamailio is kamailio who has 777 permissions in /dumps/
directory. No way.
Anyhelp would be appreciated.
Cheers,
Ricardo Domínguez
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--
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Re: [SR-Users] ACK not sent and rr-enforced
Hi Alex, As I said I was doubting about this inference, but as calls are working with other providers and I read the post I linked, I don't really know in what side the solution is. The scenario is as follows: Softphone A - providerProxy - myProxy - Softphone B Softphone A sends the invite to Softphone B through providerProxy and myProxy Softphone B sends the 200OK with CONTACT: user@softphone_B_contact_URI to myProxy myProxy sends the 200OK with CONTACT: user@softphone_B_contact_URI to providerProxy providerProxy sends the 200OK with CONTACT: user@myproxy_IP_address to Softphone A Softphone A sends ACK sip:user@myproxy_IP_address and when it arrives to myproxy it is never sent to Softphone B and there is a timeout 30 seconds later. Is there anything I can do to solve this? Thank you, Ricardo -Mensaje original- De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Alex Balashov Enviado el: martes, 15 de marzo de 2011 17:25 Para: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] ACK not sent and rr-enforced On 03/15/2011 08:28 AM, Dominguez Jover, Ricardo wrote: Should I infer IPTEL.org is not implementing SIP RFC 3261 in the right way? It seems odd to me... No, Ricardo, that is not the correct inference. First, if the ACK is an end-to-end ACK (as for a 200 OK), it is generated by the sending endpoint, and the SER proxy is not responsible for constructing it. Secondly, there are various reasons why an ACK may have a request line not equal to the Contact URI established as the dialog target, having to do with backward compatibility with RFC 2543. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
Re: [SR-Users] ACK not sent and rr-enforced
Hi, I've found this post where it says: ...the Contact header in the 200 OK and the request URI in the ACK. They MUST be the same!!!... http://www.mail-archive.com/users@lists.kamailio.org/msg00606.html Should I infer IPTEL.org is not implementing SIP RFC 3261 in the right way? It seems odd to me... Cheers, Ricardo De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Enviado el: lunes, 14 de marzo de 2011 10:58 Para: Dominguez Jover, Ricardo CC: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] ACK not sent and rr-enforced Hello, I will look over it very soon. As a hint for the future, if you catch me traveling, rar files won't work for me, use tgz or zip as they are easy to expand very easy even on web mail clients. If the trace is not big, plain text is faster or eventually use some pastebin sites out there. Cheers, Daniel On 3/10/11 1:49 PM, Dominguez Jover, Ricardo wrote: Hello Daniel, here it is. Thanks Ricardo De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Enviado el: jueves, 10 de marzo de 2011 12:49 Para: Dominguez Jover, Ricardo CC: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] ACK not sent and rr-enforced Hello, can you post the ngrep trace of such call (fron incoming invite, to the bye, taken on your server)? That will help to see what could be mismatching there. Cheers, Daniel On Thu, Mar 10, 2011 at 11:06 AM, Dominguez Jover, Ricardo djo...@umh.es wrote: Hi again, I'm still working in this issue. I've noticed that iptel proxy is writing in the ACK message the following: ACK sip:username@myproxyIP:5060;. - ACK is not sent to the client. tcheck_trans fails. If a force the transfer - t_relay do nothing while sip2sip and VoIP-Talk are writing: ACK sip:username@userprivateIP:5060; - ACK is sent to the client In both cases, contact URI sent in the 200 OK message by my proxy is the private IP address of the client sending the 200 OK, so I don't know why IPtel doesn't use it in the ACK. I find a lot of information about lost ACKs in posts, but not this particular issue. Could anyone give me some related information that can help me to solve this issue? Best regards, Ricardo Dominguez De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez Jover, Ricardo Enviado el: lunes, 07 de marzo de 2011 20:03 Para: sr-users@lists.sip-router.org Asunto: [SR-Users] ACK not sent and rr-enforced Hi everybody. I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external test accounts to check if the calls are established and the media flow is ok. When I use a sip2sip.info or VoIP Talk accounts, then all is working fine between my internal and these external accounts. But when I use a iptel.org account and this account calls to an internal account (registered with kamailio), then callee sends the 200 OK to the SIP proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy with this lines at the end of the packet: P-hint: rr-enforced\r\n P-hint: rr-enforced\r\n And my SIP proxy never resends the ACK to the callee, so the callee resends OK 200 periodically and after 32 seconds sends a BYE message and the call is finished. I've been reading posts about missing ACKs but I can't find the answer to my problem, that it seems like t_check_trans doesn´t recognize the ACK as related to a transaction. But this is only with IPTEL accounts, my proxy SIP is working with other SIP providers, so I don't know if forcing relay of every ACK packet is a good idea. Any help would be appreciated. Thanks, Ricardo ___ 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://www.asipto.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 -- Daniel-Constantin Mierla http://www.asipto.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] ACK not sent and rr-enforced
Hello Daniel, here it is. Thanks Ricardo De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Enviado el: jueves, 10 de marzo de 2011 12:49 Para: Dominguez Jover, Ricardo CC: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] ACK not sent and rr-enforced Hello, can you post the ngrep trace of such call (fron incoming invite, to the bye, taken on your server)? That will help to see what could be mismatching there. Cheers, Daniel On Thu, Mar 10, 2011 at 11:06 AM, Dominguez Jover, Ricardo djo...@umh.es wrote: Hi again, I'm still working in this issue. I've noticed that iptel proxy is writing in the ACK message the following: ACK sip:username@myproxyIP:5060;. - ACK is not sent to the client. tcheck_trans fails. If a force the transfer - t_relay do nothing while sip2sip and VoIP-Talk are writing: ACK sip:username@userprivateIP:5060; - ACK is sent to the client In both cases, contact URI sent in the 200 OK message by my proxy is the private IP address of the client sending the 200 OK, so I don't know why IPtel doesn't use it in the ACK. I find a lot of information about lost ACKs in posts, but not this particular issue. Could anyone give me some related information that can help me to solve this issue? Best regards, Ricardo Dominguez De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez Jover, Ricardo Enviado el: lunes, 07 de marzo de 2011 20:03 Para: sr-users@lists.sip-router.org Asunto: [SR-Users] ACK not sent and rr-enforced Hi everybody. I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external test accounts to check if the calls are established and the media flow is ok. When I use a sip2sip.info or VoIP Talk accounts, then all is working fine between my internal and these external accounts. But when I use a iptel.org account and this account calls to an internal account (registered with kamailio), then callee sends the 200 OK to the SIP proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy with this lines at the end of the packet: P-hint: rr-enforced\r\n P-hint: rr-enforced\r\n And my SIP proxy never resends the ACK to the callee, so the callee resends OK 200 periodically and after 32 seconds sends a BYE message and the call is finished. I've been reading posts about missing ACKs but I can't find the answer to my problem, that it seems like t_check_trans doesn´t recognize the ACK as related to a transaction. But this is only with IPTEL accounts, my proxy SIP is working with other SIP providers, so I don't know if forcing relay of every ACK packet is a good idea. Any help would be appreciated. Thanks, Ricardo ___ 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://www.asipto.com ack_trace.rar Description: ack_trace.rar ___ 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] ACK not sent and rr-enforced
Hi everybody. I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external test accounts to check if the calls are established and the media flow is ok. When I use a sip2sip.info or VoIP Talk accounts, then all is working fine between my internal and these external accounts. But when I use a iptel.org account and this account calls to an internal account (registered with kamailio), then callee sends the 200 OK to the SIP proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy with this lines at the end of the packet: P-hint: rr-enforced\r\n P-hint: rr-enforced\r\n And my SIP proxy never resends the ACK to the callee, so the callee resends OK 200 periodically and after 32 seconds sends a BYE message and the call is finished. I've been reading posts about missing ACKs but I can't find the answer to my problem, that it seems like t_check_trans doesn´t recognize the ACK as related to a transaction. But this is only with IPTEL accounts, my proxy SIP is working with other SIP providers, so I don't know if forcing relay of every ACK packet is a good idea. Any help would be appreciated. Thanks, Ricardo ___ 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] rtpproxy and connection information field
Hi all, I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every case. However I have a problem with some call. When softphone A using sip2sip.info account calls softphone B using my Kamailio server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN IP4 81.23.228.150): INVITE sip:12...@.xxx;transport=udp SIP/2.0 Record-Route: sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317 Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0 Via: SIP/2.0/UDP 192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e37bb89-1---d8754z-;rport=7964 Max-Forwards: 69 Contact: sip:54...@88.xxx.xxx.xxx:7964;transport=udp To: 2205sip:12...@.xxx From: R sip2sipsip:54...@sip2sip.info;tag=d9e99adf Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent:x Content-Length: 279 v=0 o=- 12943454020854250 1 IN IP4 192.168.xxx.xx s= c=IN IP4 81.23.228.150 t=0 0 m=audio 52854 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Then Kamilio translates c= IP address to my RTPproxy IPaddress and sends the INVITE to softphone B. After the OK softphone B sends RTP packets to the RTPproxy as specified in the c= field, however the RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send packets to IP2 (the one in the c= field?) Thanks, Ricardo Dominguez ___ 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] rtpproxy and connection information field
I have to use r flag. Sorry for my quick posting... De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez Jover, Ricardo Enviado el: martes, 01 de marzo de 2011 13:16 Para: sr-users@lists.sip-router.org Asunto: [SR-Users] rtpproxy and connection information field Hi all, I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every case. However I have a problem with some call. When softphone A using sip2sip.info account calls softphone B using my Kamailio server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN IP4 81.23.228.150): INVITE sip:12...@.xxx;transport=udp SIP/2.0 Record-Route: sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317 Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0 Via: SIP/2.0/UDP 192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e 37bb89-1---d8754z-;rport=7964 Max-Forwards: 69 Contact: sip:54...@88.xxx.xxx.xxx:7964;transport=udp To: 2205sip:12...@.xxx From: R sip2sipsip:54...@sip2sip.info;tag=d9e99adf Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent:x Content-Length: 279 v=0 o=- 12943454020854250 1 IN IP4 192.168.xxx.xx s= c=IN IP4 81.23.228.150 t=0 0 m=audio 52854 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Then Kamilio translates c= IP address to my RTPproxy IPaddress and sends the INVITE to softphone B. After the OK softphone B sends RTP packets to the RTPproxy as specified in the c= field, however the RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send packets to IP2 (the one in the c= field?) Thanks, Ricardo Dominguez ___ 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] Radius and kamailio: avp unknown attribute
Hi everybody, I have a problem with Kamalio and Radius, when a try to authenticate a user I get the following: Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: rc_avpair_new: unknown attribute 5 Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: ERROR: auth_radius [sterman.c:412]: authorization failed In fact I can't see any packet sended to the Radius Server from Kamailio. I've read several posts and I've chmod of /var/run/radius.seq, include dictionary.sip in dictionary.radius and so on. Changing if (!radius_www_authorize($td)) with if (!radius_www_authorize()) I don't get any error but and can't neither see any packet sended to the Radius Server. I've checked the connection between radius client and radius server and it's OK, I get: Received response ID 91, code 2, length = 211 Any help would be appreciated. Kind regards, Ricardo Dominguez ___ 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] Radius and kamailio: avp unknown attribute
Solved, bad dictionary Thanks ___ 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] RTP to SRTP bridge
Hi everybody, I'm trying to deploy an scenario where all calls in the LAN are encrypted with SRTP, so I've forced my sip softphones to mandatory SRTP. As in the Internet not all providers and clients have SRTP enabled, I couldn´t communicate them in this way, so I would need some kind of border element to bridge RTP and SRTP calls. Is there any bridge/proxy who can receive rtp packets and convert them into srtp and viceversa? I've been reading RTPproxy and Media proxy docs and also some forums, but I didn´t find anything useful. Regards, Ricardo Dominguez ___ 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] kamailio restart and TLS ( relay_to_tls() )
TCP ASYNC=YES fixed the problem (set_forward_no_connect() didn´t.) I don't know if you notice that TCP ASYNC in Core Cook Book v3.1 is not updated with this feature with TLS: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#tcp_async although in Kamailio 3.1 New Features DOC it is: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.1.x#asynchronous_tls Thanks a lot for your help. Best regards, Ricardo Domínguez -Mensaje original- De: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] Enviado el: martes, 21 de diciembre de 2010 20:03 Para: Dominguez Jover, Ricardo CC: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] kamailio restart and TLS ( relay_to_tls() ) Am 21.12.2010 18:46, schrieb Dominguez Jover, Ricardo: Hi again Klaus, I understand (now better) what you mean with timing parameters, I was just testing to close the first connection. The reason is because when I restart kamailio the clients I use reopen a second connection, as you said to me. So the solution to this issue could be not to open newer connection. I tested, as you said, set_forward_no_connect(); but may be not well enough. I imagine the solution goes by using it. The TCP connection should be kept alive as long as possible. If for some reason the TCP connection is lost (client crash, network failure, kamailio restart) there are two things the should be done: - the client has to create a new registration on a new TCP connection - the proxy should ignore contacts without existing TCP connections (thus use set_forward_no_connect()) About the question on making TLS connection to the clients, I'm only relaying TLS connections to the gateway, who has a certificate. I set TCP ASYNC=NO, because I had an error running TLS, as documentation says if I use TLS I have to disable asynch TCP. That was valid with 3.0 release. Since 3.1 release this is fixed and asynch mode can be used also with TLS. regards Klaus ___ 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] kamailio restart and TLS ( relay_to_tls() )
Hi everybody,
Since I implemented Kamailio 3.1 with TLS I've found a strange behavior. That
is, with some clients (Bria and Blink) registered, if I restart Kamailio, then
when the clients re-register the strange behaivour happens. This behavior
consist on receiving calls, it took about 15 seconds to receive the first tone
since the call was made.
I made the following modification in the route[Relay] config. The reason is I
wanted my gateway and Kamailio to make signaling by TLS. Without this
modification the signaling was unencrypted (SIP):
route[RELAY] {
#!ifdef WITH_NAT
if (check_route_param(nat=yes)) {
setbflag(FLB_NATB);
}
if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
route(RTPPROXY);
}
#!endif
/* example how to enable some additional event routes */
if (is_method(INVITE)) {
#t_on_branch(BRANCH_ONE);
t_on_reply(REPLY_ONE);
t_on_failure(FAIL_ONE);
}
# Se comunica con el GWa traves de TLS
if(!( ($od=~mydomain.com) ( ($rU=~[a-z]{3,20}$) ||
($rU=~^xx[0-9][0-9]$) ) ) ) {### If I'm calling a PBX extension do the
signaling by TLS with the gateway (Cisco 2811)
if (!t_relay_to_tls()) {
sl_reply_error();
}
} else if {
if (!t_relay()) {
sl_reply_error();
}
}
exit;
}
The rest of functionalities are working really fine. Any idea about what is
happening?
Cheers!
Ricardo Domínguez
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[SR-Users] append_rpid_hf() and append_hf() issues
Hi everybody,
I've been testing and setting up kamalio 3.0.3 with MySQL since a few
weeks ago. All the basic functionalities are working properly, but now I
have an issue which I'm not been able to solve by myself. I've read
documentation and posts, but no idea about what is happening. This is
the issue.
I'm trying to append Remote Party ID field to the SIP message header.
I've been trying with both append_rpid_hf() and append_hf() and I
can't see any Remote Party Header at the SIP messages sniffed with
wireshark. Here is the configuration I've tried:
#para cargar los Remote party ID de la tabla suscriber
modparam(auth_db, load_credentials, $avp(s:rpid)=rpid)
modparam(auth, rpid_avp, $avp(s:rpid))
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix,
;user=phone;party=calling;screen=yes;id-type=subscriber;privacy=off)
#Modificacion para no pedir autenticacion si viene desde un trusted
if (!allow_trusted($si, $(proto{s.toupper}))) {
if (!proxy_authorize(umh.es, subscriber)) {
proxy_challenge(, 0);
exit;
}
if (is_method(PUBLISH))
{
if ($au!=$tU) {
sl_send_reply(403,Forbidden
auth ID);
exit;
}
} else {
if ($au!=$fU) {
sl_send_reply(403,Forbidden
auth ID);
exit;
}
}
append_rpid_hf($fU,
;party=calling;id-type=subscriber;privacy=off;screen=yes);
consume_credentials();
### I also tested the following
# append_hf(Remote-Party-ID:
sip:$avp(rpid)@$avp(domain);user=phone;privacy=$avp(privacy);party=cal
ling\r\n);
# append_hf(P-Asserted-Identity:
sip:$avp(rpid)@$avp(domain)\r\n, Call-ID);
# consume_credentials();
}
}
In a similar post I've read that the user had a problem because of the
domain table. I have both the domain name and the IP of the host
defined, and I also get response with $avp(s:rpid) , when RPID field is
defined to a subscriber.
As I said I can't find the mistake, so any help would be appreciated.
Thanks in advance and best regards,
Ricardo Dominguez
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Re: [SR-Users] append_rpid_hf() and append_hf() issues
Hi Alex, I think I'm doing what you say, look at the configuration my config file: append_rpid_hf($fU, ;party=calling;id-type=subscriber;privacy=off;screen=yes); consume_credentials(); (the following lines are commented) ### I also tested the following # append_hf(Remote-Party-ID:sip:$avp(rpid)@$avp(domain);user=phone;privacy=$avp(privacy);party=calling\r\n); # append_hf(P-Asserted-Identity:sip:$avp(rpid)@$avp(domain)\r\n, Call-ID); # consume_credentials(); Is that right or you mean something like this: append_rpid_hf($avp(rpid), ;party=calling;id-type=subscriber;privacy=off;screen=yes); consume_credentials(); ? Anyway I've just tested this last configuration and I can't see Remote Party ID headers in SIP message. Kind regards, Ricardo -Mensaje original- De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Alex Balashov Enviado el: miércoles, 10 de noviembre de 2010 11:24 Para: sr-users@lists.sip-router.org Asunto: Re: [SR-Users] append_rpid_hf() and append_hf() issues Ricardo, The 'rpid' value is loaded from the 'subscriber' table as part of the authentication functions; if you consume_credentials() before it, it will be gone. You should append_rpid_hf() the RPID header first, and consume_credentials() afterward. Cheers, -- Alex ___ 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
