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37573 discussions

Websocket kamailio.cfg doubt
by Austin Einter 23 Mar '15

23 Mar '15

The README of websocket suggests to modify kamailio.cfg as below *# Handle requests within SIP dialogsroute[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } route(RELAY); } else { if ( is_method("ACK") ) {...* But the original script was *# Handle requests within SIP dialogsroute[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit;}* Now I have to include the websocket README suggested script in original script. I am not finding logical way where shall I include websocket part. Can somebody please help me. Thanks Austin

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Kamailio sending NOTIFY messages to private IP
by Olli Attila 23 Mar '15

23 Mar '15

Hello, Im running kamailio 4.2.3 and I'm having troubles with reporting user presence data from Kamailio server to a registered UA behind nat. The NOTIFY message is being sent from Kamailio to the UA after receiving PUBLISH, but because the UA is behind nat the NOTIFY message is being sent to the UAs private (=unroutable) ip and therefor never reaching the real UA. My server is using nat helping modules so that REGISTER messages and INVITE messages are handled ok even if they are originated behind nat. Any suggestions how to solve this and make the NOTIFY messages to be sent via routabe ip addresses? my kamailio.cfg PRESENCE routing: #!ifdef WITH_PRESENCE 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; #!endif if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } -- Olli

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Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: <sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes> --->Kamailio route From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2>;tag=3b27a2a409 Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2>;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as00936a6d To: <sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2>;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- 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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- 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 BQ_END -- 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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- 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

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Re: [SR-Users] remove_hf_value
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Daniel, I end up insert routes like this. And right now on client side I see only public ip route and private side only private ip routes. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("NOTIFY")) { if(is_ip_rfc1918($si)) { xlog("L_INFO", "[$rm], Present Route: $(route_uri{uri.host})\n"); remove_hf("Route"); insert_hf("Record-Route: <"sip" + $sel(cfg_get.kamailio.bindpublicip) + ":" $sel(cfg_get.kamailio.bindpublicport);transport=tcp;lr=on;ftag=$ft;nat=yes>\r\n"); $fs = "tcp" + ":" + $sel(cfg_get.kamailio.bindpublicip) + ":" + $sel(cfg_get.kamailio.bindpublicport); } else { xlog("L_INFO", "[$rm], Present Route: $(route_uri{uri.host})\n"); remove_hf("Route"); insert_hf("Record-Route: <"sip" + ":" $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes>\r\n"); $fs = "udp" + ":" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport); } } } } And for initial SUBSCRIBE INVITE if (is_method("INVITE|SUBSCRIBE")) { remove_hf("Route"); insert_hf("Record-Route: <"sip" + ":" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes>\r\n"); #record_route(); } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 18, 2015 4:15:28 AM Subject: Re: [SR-Users] remove_hf_value Hello, but the INVITE request goes out with 2 Record Route headers, if you get them in the 200ok. Not sure how it is going to work overall, you will have to manually set local sockets for requests within dialog, because you are doing also trasport layer gatewaying (tcp-udp). Anyhow, as a solution, maybe you should just not be using record_route() function and just add the header with insert_hf() for the invite, remove the Record-Route header for 200ok with remove_hf() and insert a new one again with insert_hf() If you know the local address (IP/port/transport) you want to put in the Record-Route, you can build the header as: insert_hf("Record-Route: <sip:_LOCAL_IP_:_LOCAL_PORT_;transport=_LOCAL_TRANSPORT_;lr=on;ftag=$ft;nat=yes> \r\n"); Requests within the dialog can be handled with loose_route(), but again, you have to do more in config for selecting the local socket to send to next hop. Cheers, Daniel On 16/03/15 13:35, Slava Bendersky wrote: Hello Daniel, I want do clean up on routes from first SUBSCRIBE, that when traffic goes to public side of kamailio will be only public route and when traffic is internal the Record-Route will stay internal only. In SDP I see like this. I tired double routes set 0 in rr, but didn't clean enough. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TCP 192.168.88.252:5062;received=client_pub_ipbranch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> On public side remote this Record-Route Record-Route: <sip:kamailio_pub_ip:5084;transport=tcp;r2=on;lr=on;ftag=a185d974ec;nat=yes> On private side remote this Record-Route From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec To: <sips:10101@networklab.ca> ;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060> ;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 16, 2015 7:01:54 AM Subject: Re: [SR-Users] remove_hf_value Hello, can you give an example of incoming header and how you want to be after modification? It is not clear for me what end result you want/expect. Cheers, Daniel On 14/03/15 06:41, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Here updated version. Trying remove Record-Route after the coma. My log shows Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Inbound connection from <Client Public IP> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: <script>: <null> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Route for removal: '<null>' route snippet. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("SUBSCRIBE")) { if(is_ip_rfc1918($si)) { xlog( "$sel((a)hf_value.Record_Route[-1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[-1])'\n"); remove_hf_value("Record-Route[-1]"); } else { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } } return; } Slava, From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:00:44 PM Subject: [SR-Users] remove_hf_value Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- 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

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Presence data to users behind NAT
by Olli Attila 22 Mar '15

22 Mar '15

Hello, Im running kamailio 4.2.3 and I'm trying to build a system where I could have the following features: SIP proxy, RTP proxy, Presence & XCAP service for IM messaging. I have a problem with the Presence service with users that are located behind nat. When Kamailio sends presence info to a watcher with sip NOTIFY message, the destaination ip for the pacage seems to be an un-routable (=private) ip, and so the watcher never receives the NOTIFY message making the watchers sip client unaware of the other users presence. Any suggestions how to make Kamailio handle the watchers behind nat so that the NOTIFY messages will get delivered succesfully? I tried the scenario with a client with public ip and the presence works fine. my kamailio.cfg: http://tny.cz/7d85773c -- Olli

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Kamailio with websocket as transport
by Austin Einter 22 Mar '15

22 Mar '15

Dear All I have a sip user agent. Today it works fine with Kamailio using tcp as tranasport. Now I am planning to support websocket in my sip ua. My question is does Kamailio support websocket transport. If so how can I configure it or while building kamailio do I need to make some flags on. I am using latest Kamailio 4.2.3 (version: kamailio 4.2.3 (x86_64/linux)). Thanks Austin

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Hardware config to run 1000 Kamailio TLS users
by elektau＠yandex.ru 20 Mar '15

20 Mar '15

Hi All, 1. What are resource requirements for running Kamailio server with TLS module active for 1000 registered users (concurrent calls) when running on VPS? In other words, what amount of RAM and processor power required? 2. What is the maximum number of Kamailio TLS users/concurrent calls can be made for the below default values? # Amount of shared and private memory to allocate # for the running Kamailio server (in Mb) SHM_MEMORY=64 PKG_MEMORY=8 Thanks.

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Kamailio hardware requirements
by Mickael Marrache 20 Mar '15

20 Mar '15

Hi, We are currently deploying an entire architecture composed of load balancers, proxies and media relays. All the components except the media relays are Kamailio instances. The media relays are RTP proxy instances. We are trying to determine the hardware requirements for the different servers. We will start with an architecture composed of: . 2 load balancers (one kept as backup to ensure availability) . 2 proxies (load is balanced over the 2 using the DISPATCHER module) . 2 media relays We are trying to determine the recommended hardware for the different components taking into account we will need to serve around 400000 users. How many registrations per second can Kamailio support? How many concurrent calls can Kamailio support? What is the recommended number of children processes? What is the recommended size for PKG memory? What is the recommended size for SHM memory? Thanks for your help, Mickael

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auth_diameter module
by Grant Bagdasarian 19 Mar '15

19 Mar '15

Hello, Does Kamailio still support authentication using the auth_diameter module? There is a note on the modules page stating the module is now obsolete. "NOTE: diameter support was developed for DISC (DIameter Server Client project at http://developer.berlios.de/projects/disc/). This project seems to be no longer maintained and DIAMETER specifications were updated in the meantime. Thus, the module is obsolete and needs rework to be usable with opendiameter or other DIAMETER servers." Is the page referring to this module or the DISC? Are there perhaps any other ways to do authentication with Kamailio using diameter?

2 1

RTPProxy issue?
by Igor Potjevlesch 19 Mar '15

19 Mar '15

Hello, I discovered an issue related to the handling of "timestamp" and/or "Marker bit" with rtpproxy (I use the latest Extension 20081224). The call-flow is the following: one UA places a call to A and put this call on hold. Then, the same UA call another number B. Individual streams are ok. When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with a huge timestamp and the Marker bit set to "True". Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. B to C is still working. Anyone have an idea? Regards, Igor.

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