[SR-Users] LinuxTag Today: Workshop build your own UC service with Kamailio and Jitsi
Hello, short message to let you know that today at 12:00 I will do a practical workshop at LinuxTag in Berlin, about how to build your own unified communication service using kamailio as server and jitsi as client. If you visit the event, you can join the workshop in room New York 2.1, more details at: http://www.linuxtag.org/2012/en/program/project-workshops/details.html?no_cache=1talkid=705 Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] vsf value in Record-Route header
Hello, vsf header is added if you use uac module and uac_replace_from() function. Is it the case? Cheers, Daniel On 5/22/12 10:03 PM, Robert R wrote: Hi, / / / / We are facing a problem in our opener proxy receiving: /Message too big error./ Looking at the INVITE message sent by opener I notice the following in the record-route header: Record-Route: sip:10.10.10.10;lr=on;ftag=5114A38-F80;vsf=AAA- Is this a valid value for vsd? Can we correct the value? Thanks, R ___ 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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Question about more complicated rtpproxy scenario
Hi! Some hints: Do not try to put several RTP proxies in a row - only one instance per call should be used. Consider this simple example (I do not know the details of your network): - internal LAN: IPv4 only - external: IPv4 + IPv6 - the SBC handles only int-ext calls (no int-int and no ext-ext) Then create 2 rtpproxy sets. rtpproxy's first interface is the internal, the second is the external: 1) IPv4(int)-IPv4(ext): rtpproxy -l int.ip.add.ress/ext.ip.add.ress [-6 addr1[/addr2]] 2) IPv4(int)-IPv6(ext): rtpproxy -l int.ip.add.ress -6 /ext.ipv6.add.ress For requests from extern-v4 to intern use: set_rtp_proxy_set(1); rtpproxy_manage(ei) and for the respective responses use: set_rtp_proxy_set(1); rtpproxy_manage(ie) For requests from intern to extern-v4 use: set_rtp_proxy_set(1); rtpproxy_manage(ie) and for the respective responses use: set_rtp_proxy_set(1); rtpproxy_manage(ei) For requests from extern-v6 to intern use: set_rtp_proxy_set(2); rtpproxy_manage(ei) and for the respective responses use: set_rtp_proxy_set(2); rtpproxy_manage(ie) For requests from intern to extern-v6 use: set_rtp_proxy_set(2); rtpproxy_manage(ie) and for the respective responses use: set_rtp_proxy_set(2); rtpproxy_manage(ei) Of course you need some ways to find out if the target is IP4 or IP6 before calling set_rtp_proxy_set(). If you use IPv6 also internally, then add another proxy set with internal and external IPv6 address. regards Klaus On 23.05.2012 01:53, Lukas Macura wrote: Hi to all, please could somevody help me with more complicated rtpproxy scenario? I still cannot understand how to make it working. We have SBC based on kamailio 3.2.3. Everything seems to work until I want to translate all SDP messages by SBC IP address (for both sides). Moreover, I need to translate ipv4 to ipv6 and vice versa too. I found some example script how to achieve ipv4 to ipv6 and vice versa and I am tryinf to use it. It is relatively easy to translate all RFC addresses to one public IP, but I need to translate even our public IPS into SBC IP. I made some rtpproxy sets: modparam(rtpproxy, rtpproxy_sock, 1 == udp:127.0.0.1:7722) (EXTRA_OPTS=-l SBC_IP) for internal net to external modparam(rtpproxy, rtpproxy_sock, 2 == udp:127.0.0.1:7723) (EXTRA_OPTS=-l /SBC_IP -6 SBC_IP6) for ipv4 to ipv6 modparam(rtpproxy, rtpproxy_sock, 3 == udp:127.0.0.1:7724) (EXTRA_OPTS=-l SBC_IP -6 /SBC_IP6) for ipv6 to ipv4 modparam(rtpproxy, rtpproxy_sock, 4 == udp:127.0.0.1:7725) (EXTRA_OPTS=-l SBC_IP/SBC_IP) for external to internal but I cannot make it working. Either there are problems with incorrect port 0 in reply from rtp proxy or SDP gets malformed because of more times rtpproxy_manage(). Please can you help me with some basic hint, where into routing logic put rtpproxy_manage() and set_rtpproxy_set() ? Moreover, how to check if rtpproxy was already forced? I could not find any example of forcing RTP proxy for all calls for both sides. I need this because entire RTP traffic has to go across SBC. And my internal IPs are not private. Thank you for any hint, Lukas Macura ___ 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] rtpproxy and timeout notification
Hello all, I see the new timeout notification feature from the rtpproxy when no voice traffic is detected. I have 2 questions regarding this timeout notification: 1) When a timeout is detected and kamailio send the BYE is that possible to manipulate the BYE before he is sent to add maybe an extra header that indicate that the call was stopped due to a timeout issue ? 2) If a user put the call on hold (with a=recvonly or maybe old way IP=0.0.0.0.0) what will happens (the music on hold is not generated by rtpproxy but by another server) ? he will detect this as a timeout ? Is that possible to indicate to rtpproxy that for a moment stream will be only in one way ? Best regards Laurent ___ 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] Sending MSRP message to kamailio/msrprelay causes 501 error
Hi, We are developing a Java based SIP gateway implementation and are performing some integration tests on a kamailio/msrprelay server (plugin from www.msrprelay.org). The SIP signaling part works so far, but when sending a MSRP message, the client receives a 501 Unknown method error. However, it seems the request does not reach the MSRP module on the server as there is no corresponding trace log on the server side. On Protocol level, everything looks fine so far, the sequence executed is as follows: --SIP REGISTER -- SIP/2.0 401 Unauthorized --SIP REGISTER --SIP/2.0 200 OK --INVITE --SIP/2.0 407 Proxy Authentication Required --ACK --INVITE --SIP/2.0 180 Ringing --SIP/2.0 180 Ringing --SIP/2.0 200 OK --ACK The INVITE is processed with the following SDP packet in the request: v=0 o=- 1337788359935 1337788359935 IN IP4 localhost s=MSRP Session c=IN IP4 localhost t=0 0 m=message 2855 TCP/MSRP * a=path:msrp://localhost:2855/U3FuvW00m62UY0020H71;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:active And response: v=0 o=- 3546777215 3546777216 IN IP4 46.105.78.68 s=sipsimple 0.20.0 c=IN IP4 46.105.78.68 t=0 0 m=message 57233 TCP/MSRP * a=path:msrp://msrphost:2855/5AUNTOt7XlFsK7VuT4SzETEzMzc3ODg0MTUuMDcyOjQ2LjEwNS43OC42OA==;tcp msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:passive But when sending the MSRP message, 501 is returned: MSRP f0CHA1TD SEND To-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp From-Path: msrp://localhost:52673/b10XadK9;tcp Message-ID: 13377883623135c260a3568cc Byte-Range: 1-*/14 Hello Francois ---f0CHA1TD$ MSRP f0CHA1TD 501 Unknown method: SEND To-Path: msrp://localhost:52673/b10XadK9;tcp From-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp ---f0CHA1TD$ When testing with two sip-simple client endpoints, MSRP messaging works fine, but on protocol traces we cannot identify any differences between sip-simple communication flows and the ones we are implementing. Does anyone have a hint on what is wrong in this message flow? Any advice is appreciated. Thanks, 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
Re: [SR-Users] kamailioi is not relaying BYE message to UAC
Thanks for the reply.
We have already came to the same conclusion by some testing in our
lab. It seems its a bug in provider which not constructing BYE message
properly.
But i'm interested in if its possible to detect the fault in this BYE
and construct a new one and then relay it to the UAC.
I mean can i do this :
contact-header = INVITE's contact-header
if (contact-header != BYE's ruri)
{
construct BYE message with contact header
t_relay()
}
Cheers
aft
Hi aft,
I think to do what you want you can save Contact field + callid pair
taken from the INVITEs that comes from your side.
Then when BYE comes from such provider you should find a correct Contact
for that call-id and if it exists and not equal to R-URI you can rewrite it.
For example to write to R-URI value from $var(correct_ruri) you can use
*$ru = $var(correct_ruri);* statement.
What about a place where contact can be saved, i think htable will be
the nice one.
Please pay attention to autoexpire parameter of htable module. You
should take care about the staled records to avoid memory usage problems.
Cheers,
Vitaliy Aleksandrov
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Re: [SR-Users] Sending MSRP message to kamailio/msrprelay causes 501 error
Hello, kamailio has an embedded msrp relay functionality. You have to load and configure msrp module -- it is new in upcoming release (3.3.0, planned to be out about mid of June): http://kamailio.org/docs/modules/devel/modules/msrp.html Have you tried with it as well? I developed the current version, but had no good clients I could test with at that time, so I used mainly network sending tools to simulate msrp traffic. Cheers, Daniel On 5/24/12 1:27 PM, Reichert Alexander wrote: Hi, We are developing a Java based SIP gateway implementation and are performing some integration tests on a kamailio/msrprelay server (plugin from www.msrprelay.org). The SIP signaling part works so far, but when sending a MSRP message, the client receives a 501 Unknown method error. However, it seems the request does not reach the MSRP module on the server as there is no corresponding trace log on the server side. On Protocol level, everything looks fine so far, the sequence executed is as follows: --SIP REGISTER -- SIP/2.0 401 Unauthorized --SIP REGISTER --SIP/2.0 200 OK --INVITE --SIP/2.0 407 Proxy Authentication Required --ACK --INVITE --SIP/2.0 180 Ringing --SIP/2.0 180 Ringing --SIP/2.0 200 OK --ACK The INVITE is processed with the following SDP packet in the request: v=0 o=- 1337788359935 1337788359935 IN IP4 localhost s=MSRP Session c=IN IP4 localhost t=0 0 m=message 2855 TCP/MSRP * a=path:msrp://localhost:2855/U3FuvW00m62UY0020H71;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:active And response: v=0 o=- 3546777215 3546777216 IN IP4 46.105.78.68 s=sipsimple 0.20.0 c=IN IP4 46.105.78.68 t=0 0 m=message 57233 TCP/MSRP * a=path:msrp://msrphost:2855/5AUNTOt7XlFsK7VuT4SzETEzMzc3ODg0MTUuMDcyOjQ2LjEwNS43OC42OA==;tcp msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:passive But when sending the MSRP message, 501 is returned: MSRP f0CHA1TD SEND To-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp From-Path: msrp://localhost:52673/b10XadK9;tcp Message-ID: 13377883623135c260a3568cc Byte-Range: 1-*/14 Hello Francois ---f0CHA1TD$ MSRP f0CHA1TD 501 Unknown method: SEND To-Path: msrp://localhost:52673/b10XadK9;tcp From-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp ---f0CHA1TD$ When testing with two sip-simple client endpoints, MSRP messaging works fine, but on protocol traces we cannot identify any differences between sip-simple communication flows and the ones we are implementing. Does anyone have a hint on what is wrong in this message flow? Any advice is appreciated. Thanks, 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Hop by hop CANCEL, wrong to header
Hello, On 5/23/12 5:22 PM, Vitaliy Aleksandrov wrote: Hi all, I have a question about CANCEL message processing. My call sceraio: When an INVITE request comes I need to rewrite a domain part of the several headers(to, from, contact, SDP ip) according to the outgoing interface. I can do it before t_relay(), but when destination user has more then one locations and they are reachable through different interfaces all forked INVITEs will have the same domain. To avoid that problem i have tried to move rewriting part(subst from the textops module) to the branch_route. Unfortunately that solution didn't helped me, because To header of the outgoing CANCEL messages is wrong (unchanged. as it was at the moment when transaction was created by the t_relay() ). As i understood from the documentation and mailing list kamailio builds CANCEL based on outgoing INVITE. I did a small research and found that Kamailio really takes outgoing INVITE from each branch (invite_transaction_cell - uac[b_id].request.buffer) and builds CANCEL (build_local_reparse() from tm/t_msgbuilder.c) based on it. But kamailio does an exception for the To header which is described below: ***cancel_branch* function (from t_cancel.c) calls *build_local_reparse*() and fills one of the parameters with a pointer to unmodified To header. *build_local_reparse*() uses received To header to construct outgoing CANCEL. I have changed: cancel = build_local_reparse(t, branch, len, CANCEL, CANCEL_LEN, t-to, reason); to: cancel = build_local_reparse(t, branch, len, CANCEL, CANCEL_LEN, NULL, reason); and it works for me now. It looks like if E2E_CANCEL_HOP_BY_HOP *e2e_cancel()* (t_fwd.c) will call *e2e_cancel_branch()* which works as i want instead of *cancel_branch()*, but it is just my assumption. do you mean if E2E_CANCEL_HOP_BY_HOP is defined? I quick grep at this time showed it is defined in t_fwd.h... What is the version you are using? Cheers, Daniel Why does kamailio generate CANCEL requests in such a way ? Did i miss something from the RFC3261 or kamailio documentation ? Thanks in advance for any help ! ___ 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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] TLS config file
Hello, if there is none matching the ip address of the interface, the default one is used. So you can define one like [server:127.0.0.1:5061] just to be valid for connections coming on loopback interface. You can replace the ip and the port to fit your actual wan or lan address. Cheers, Daniel On 5/22/12 4:20 PM, Bruno Bresciani wrote: The parameter [server:default] of TLS config file must be declared? I want specify two different interfaces (WAN and LAN) and I don't know which interface will be used as default by TLS module... Someone know some information about my question? Best Regards ___ 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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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
