[SR-Users] many-to-one network topology hide with TOPOH
Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only the IPs of all VIA and RR but also the amount of hops - so to map 4 VIAs to 1 different VIA etc. The idea is to reduce the size of packets as some of my subscribers suffers from packets fragmentation. Do you think it's possible now or maybe easy to implement? Best Regards, Maciej ___ 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] many-to-one network topology hide with TOPOH
Hello, at this moment the topoh module is encoding the content of exiting headers and decodes them on the other direction. Cheers, Daniel On 05/06/14 09:17, Maciej Marczyński wrote: Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only the IPs of all VIA and RR but also the amount of hops - so to map 4 VIAs to 1 different VIA etc. The idea is to reduce the size of packets as some of my subscribers suffers from packets fragmentation. Do you think it's possible now or maybe easy to implement? Best Regards, Maciej ___ 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] check for the number of open branches in Reply
Hello, On 04/06/14 11:49, Sebastian Damm wrote: Hi, I have a scenario where I want to send a custom SIP package whenever a call is successfully established or definitely missed. This works pretty well, except when a user has multiple devices online. I need to send exactly one packet. When initiating the call I set a flag, in the reply route I check for that flag, and on the first 4xx or 2xx response the packet gets sent out. After that I reset the flag, so no other packet is generated for another response coming in. Now when a user has two devices and rejects the call on the first one but accepts it on the second, I see this call as missed, not answered. So I need to use the first 2xx response or the last 4xx response. For that I need to find out how many branches are open at the moment. But I'm missing something. I registered two devices on the same AOR. I tried accessing $branch(count) in the onreply route, but it is always 0. $br and $bR are . Does anyone have an idea, how to get that information in the reply_route? try to save the number of branches in an avp before t_relay() in request_route. Cheers, DAniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] CANCEL requests
Hi all, just a simple question: which is the timer that triggers if no reply is received for the current transaction and makes Kamailio send a CANCEL request? Andrea ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Problem with MSRP AUTH
Hi, I am trying to authentication MSRP connection using the example code of msrp event route in module documentation here, http://kamailio.org/docs/modules/4.1.x/modules/msrp.html#idp119248 -- ... } else if ($msrp(method)=="AUTH") { ... if (!pv_www_authenticate("WEBRTC_SIP_REALM", "$var(passwd)", "0", "$msrp(method)")) { if (auth_get_www_authenticate("WEBRTC_SIP_REALM", "0", "$var(wauth)")) { xlog("L_INFO", "Generated www authenticate header for MSRP is [$var(wauth)] \n"); msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Internal Server Error"); }; exit; }; ... -- However i see in logs following error message and authentication fails, -- ERROR: auth [auth_mod.c:690]: pv_www_authenticate2(): failed to get method value -- Which is indicates that value of $msrp(method) is null. However, as you can see in example code in URL provided above we have an IF condition which explicitly checks $msrp(method) == "AUTH". For the sake of testing i even replaced the variable with actual string value, and still get the same error. So i am guessing it is a bug. What do you guys suggest? Kamailio: v4.1.3 (i386/linux) 236326 MSRP Lib: Crocodile MSRP - v1.0.0 Thank you. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] CANCEL requests
Hello, depending if there is a provisional reply or not, it is fr_inv_timer or fr_timer -- see the readme of tm module for more details. Cheers, Daniel On 06/06/14 10:40, Andrea Meroni wrote: Hi all, just a simple question: which is the timer that triggers if no reply is received for the current transaction and makes Kamailio send a CANCEL request? Andrea ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Problem with MSRP AUTH
Hello, is there any other error message before the one from auth module? Cheers, Daniel On 06/06/14 10:50, Muhammad Shahzad wrote: Hi, I am trying to authentication MSRP connection using the example code of msrp event route in module documentation here, http://kamailio.org/docs/modules/4.1.x/modules/msrp.html#idp119248 -- ... }else if ($msrp(method)=="AUTH") { ... if (!pv_www_authenticate("WEBRTC_SIP_REALM", "$var(passwd)", "0", "$msrp(method)")) { if (auth_get_www_authenticate("WEBRTC_SIP_REALM", "0", "$var(wauth)")) { xlog("L_INFO", "Generated www authenticate header for MSRP is [$var(wauth)] \n"); msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Internal Server Error"); }; exit; }; ... -- However i see in logs following error message and authentication fails, -- ERROR: auth [auth_mod.c:690]: pv_www_authenticate2(): failed to get method value -- Which is indicates that value of $msrp(method) is null. However, as you can see in example code in URL provided above we have an IF condition which explicitly checks $msrp(method) == "AUTH". For the sake of testing i even replaced the variable with actual string value, and still get the same error. So i am guessing it is a bug. What do you guys suggest? Kamailio: v4.1.3 (i386/linux) 236326 MSRP Lib: Crocodile MSRP - v1.0.0 Thank you. ___ 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] CANCEL requests
Ok, thank you very much Andrea On Fri, Jun 6, 2014 at 10:01 AM, Daniel-Constantin Mierla wrote: > Hello, > > depending if there is a provisional reply or not, it is fr_inv_timer or > fr_timer -- see the readme of tm module for more details. > > Cheers, > Daniel > > > On 06/06/14 10:40, Andrea Meroni wrote: > > Hi all, > > just a simple question: which is the timer that triggers if no reply is > received for the current transaction and makes Kamailio send a CANCEL > request? > > Andrea > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://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 > > ___ 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] Problem with MSRP AUTH
Nope, just WS handshake message, INFO:
Re: [SR-Users] Problem with MSRP AUTH
In the code I couldn't spot what can be wrong at a quick look. Can you send the log messages with debug=3 in kamailio.cfg? Cheers, Daniel On 06/06/14 11:55, Muhammad Shahzad wrote: Nope, just WS handshake message, INFO:
Re: [SR-Users] Problem with MSRP AUTH
OK sure. I will provide it tonight. Thank you. On Fri, Jun 6, 2014 at 2:48 PM, Daniel-Constantin Mierla wrote: > In the code I couldn't spot what can be wrong at a quick look. > > Can you send the log messages with debug=3 in kamailio.cfg? > > Cheers, > Daniel > > > On 06/06/14 11:55, Muhammad Shahzad wrote: > > Nope, just WS handshake message, > > INFO:
[SR-Users] SDP and NAT
Hey all, I have a Kamailio box functioning as a proxy. When some UAs send an INVITE the SDP has their private IP (c=IN IP4 10.x.x.x). Kamailio is passing this on without changing it to the proper public IP. What's the best way to rewrite that? I'm using a config based on the default one. ___ 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] SDP and NAT
You can rewrite it using fix_nated_sdp(), but realistically, you're going to want to use rtpproxy instead, whose module will rewrite it. On 06/06/2014 11:40 AM, Marc Soda wrote: Hey all, I have a Kamailio box functioning as a proxy. When some UAs send an INVITE the SDP has their private IP (c=IN IP4 10.x.x.x). Kamailio is passing this on without changing it to the proper public IP. What's the best way to rewrite that? I'm using a config based on the default one. ___ 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 -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906 ___ 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] Webrtc: Don't catch 488 between JSSIP and SIP UA's
Hi all, Another attempt, After doing some tests, I saw that one of the problems was that was necessary to comment the following lines within the deffinition of the RELAY route: # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. #if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { #if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); #} At this time, all the features of my old configuration are working fine, and I can make good calls beetween standard SIP UA's and JSSIP UA's, originationg the call from both sides. I'm Using Cisco SPA3000 GW, and Twinkle and SJ Phone softphones. But now, problems are still present with calls between JS SIP UA's. When the B side accepts the calls, Kamailio sends OK to A side, but Offering RTP/AVP instead of RTP/SAVFP, and the call is been rejected because of BAD Media Description. Can anybody help me with this issue, please?. Here you are a snippet of my config, wich is based in the standard one, mixed with the example of Carlos Ruiz Diaz: ### Routing Logic # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { #!ifdef WITH_WEBSOCKETS if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } #!endif # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting #!ifdef WITH_CPL if(!cpl_run_script("incoming","is_stateful")) { # script execution failed t_reply("500","CPL script execution failed"); }; #!endif } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #if (!is_method("INVITE")) { #route(RELAY); #exit; #} if (!is_method("INVITE")) { route(RELAY); exit; } # user location service route(LOCATION); } #!ifdef WITH_WEBRTCGW route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); rtpproxy_manage("froc-sp"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } #!endif route[RELAY] { #!ifdef WITH_WEBRTCGW route(SETUP_BY_TRANSPORT); #!endif # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. #if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { #if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); #} if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null)
Re: [SR-Users] Problem with MSRP AUTH
I have sent you logs to your private email separately, did you get them? Thank you. On Fri, Jun 6, 2014 at 3:48 PM, Muhammad Shahzad wrote: > OK sure. I will provide it tonight. > > Thank you. > > > > > On Fri, Jun 6, 2014 at 2:48 PM, Daniel-Constantin Mierla < > mico...@gmail.com> wrote: > >> In the code I couldn't spot what can be wrong at a quick look. >> >> Can you send the log messages with debug=3 in kamailio.cfg? >> >> Cheers, >> Daniel >> >> >> On 06/06/14 11:55, Muhammad Shahzad wrote: >> >> Nope, just WS handshake message, >> >> INFO: