- sr-users - kamailio.org

kamailio command line
by betul.cerit＠netgsm.com.tr 28 Apr '14

28 Apr '14

Hi to all, I’m using “kamailio” command to see alerts,notices and warnings but it also starts a new kamailio process.Is there a way to see command line without starting a new kamailio process? I need something just like asterisk –rvvvvvvvvvvv in asterisk. Thanks in Advance.

2 2

No secure attributes from rtpengine in SRTP/RTP bridge mode
by Alexey Rybalko 28 Apr '14

28 Apr '14

Hello! I have been experimenting with drop-in replacement of old "rtpproxy-ng" module with new "rtpengine". Wondering what is wrong in my configuration: there are no security attributes in rtpengine answer on RTP/SAVPF offer. Neither "fingerprint" (DTLS) nor "crypto" (SDES). I used Firefox 29 during this test. 1. Here's original offer : INVITE sip:user5@..... SIP/2.0 > Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 > Max-Forwards: 69 > To: <sip:user5@......> > From: "user4" <sip:user4@......>;tag=dvf1co8urv > Call-ID: phmq9o62cv21timhfnpf > CSeq: 4233 INVITE > Proxy-Authorization: Digest algorithm=MD5, username="user4", > realm="......", nonce="U1o8H1NaOvP3wxegsYCKOJX7S7DV/r1N", uri="sip:user5@......", > response="44e7b16c55d3237f63e04b3c0b194f45" > Contact: <sip:user4@ > ......;gr=urn:uuid:c193bcd4-aa2e-47ef-a106-22e60f5fde9e;ob> > Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE > Content-Type: application/sdp > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.7 > Content-Length: 607 > > v=0 > o=Mozilla-SIPUA-29.0 15825 0 IN IP4 0.0.0.0 > s=SIP Call > t=0 0 > a=ice-ufrag:2102f082 > a=ice-pwd:8733e5248a7fb087b40ea45b3ca6f634 > *a=fingerprint:sha-256 > 32:AA:85:DB:D8:3C:E6:C3:46:07:11:9E:9F:54:B9:42:7F:5C:37:5F:9D:D1:AD:19:22:A3:AC:9C:6C:A5:A6:CD* > m=audio 62290 *RTP/SAVPF* 109 0 8 101 > c=IN IP4 ..... > a=rtpmap:109 opus/48000/2 > a=ptime:20 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > ..... > Here's the snippet for translation SRTP-RTP: rtpengine_offer("force trust-address symmetric replace-origin replace-session-connection ICE=force_relay *RTP/AVP*"); 2. Here's final answer (from rtpengine): SIP/2.0 200 OK > Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 > Record-Route: <sip:......;lr;nat=yes> > Record-Route: <sip:.....:15060;transport=udp;lr;ovid=3207d8cd> > Record-Route: <sip:95f6551e81@.......:10080;transport=ws;lr;ovid=3207d8cd> > Contact: <sip:user5@......1:49362> > To: <sip:user5@.....>;tag=4d436110 > From: "user4"<sip:user4@....>;tag=dvf1co8urv > Call-ID: phmq9o62cv21timhfnpf > CSeq: 4233 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces, eventlist > > User-Agent: X-Lite release 4.5.4 stamp 70864 > Content-Length: 432 > > v=0 > o=- 1398422264455879 3 IN IP4 ..... > s=X-Lite 4 release 4.5.4 stamp 70864 > c=IN IP4 ...... > t=0 0 > m=audio 30002 *RTP/SAVPF* 109 0 8 101 > a=rtpmap:109 opus/48000/2 > a=fmtp:109 useinbandfec=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=rtcp:30003 > .... > Error from JsSIP: {name: "INTERNAL_ERROR", message: "Could not negotiate answer SDP; cause = > *NO_DTLS_FINGERPRINT*", __exposedProps__: Object} > Here's the snippet for translation RTP-SRTP: rtpengine_answer("force trust-address symmetric replace-origin replace-session-connection rtcp-mux-demux ICE=force *RTP/SAVPF* "); There was another test with Chrome 34 with the same result. Offer: INVITE sip:user5@.... SIP/2.0 Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 Max-Forwards: 69 To: <sip:user5@....> From: "user4" <sip:user4@....>;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: 957 INVITE Proxy-Authorization: Digest algorithm=MD5, username="user4", realm=".....", nonce="U1pKZ1NaSTtLo2vjK9TGyBu6Axb+EtyN", uri="sip:user5@...", response="98f6275d3636664b611ff1411af982af" Contact: <sip:user4@....;gr=urn:uuid:8cd4e797-7314-485f-b191-4a15a6581c42;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 1586 v=0 o=- 7510391807340598328 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS ErnvtgCDe9aR6LWxNRgT83r0mxtyAV87LUxT m=audio 51672 *RTP/SAVPF *111 103 104 0 8 106 105 13 126 c=IN IP4 10.61.2.151 a=rtcp:51672 IN IP4 .... ...... a=ice-ufrag:EMn7uHfSS7ulRGU2 a=ice-pwd:FskTdhj7qT6ELP7uTIb+gquQ a=ice-options:google-ice *a=fingerprint:sha-256 46:6E:E0:18:4A:C5:06:A8:26:85:ED:FE:16:C1:86:5E:8D:BC:4D:D9:F2:1A:75:81:A1:A7:CE:5A:79:4D:B7:22*a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux *a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:R6qaiU7Cm471zNF6f3Q87TyXbHjEt/VhLgUgY2ZZ a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lch+mfN/hi9QmseWu+ss1M2vA8mwRh8GQYChaJvc* a=rtpmap:111 opus/48000/2 .... Answer: SIP/2.0 200 OK Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 > Record-Route: <sip:...;lr;nat=yes> Record-Route: > <sip:....:15060;transport=udp;lr;ovid=3207d8cd> Record-Route: > <sip:712a450958@...:10080;transport=ws;lr;ovid=3207d8cd> Contact: < > sip:user5@10.61.2.151:49362> To: <sip:user5@....>;tag=b8c8ee57 From: > "user4"<sip:user4@...>;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: > 957 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, > MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, > eventlist User-Agent: X-Lite release 4.5.4 stamp 70864 Content-Length: 432 > v=0 o=- 1398425917116411 3 IN IP4 ... s=X-Lite 4 release 4.5.4 stamp 70864 > c=IN IP4 .... t=0 0 m=audio 30006 *RTP/SAVPF* 111 0 8 126 a=rtpmap:111 > opus/48000/2 a=fmtp:111 useinbandfec=1 a=rtpmap:126 telephone-event/8000 > a=fmtp:126 0-15 a=sendrecv a=rtcp:30007 .... > Error is JsSIP: Failed to set remote answer sdp: Called with a SDP without crypto enabled > Any help on this issue would be greatly appreciated! best regards, Alexey

3 7

Double attempt to record-route
by Juha Heinanen 27 Apr '14

27 Apr '14

I tried to call record_route_preset() in branch route in order to add contact specific RR headers when request is forked to more than one contact. What happened was that I got error message: Sep 26 17:03:47 wheezy2 /usr/sbin/sip-proxy[2436]: ERROR: rr [rr_mod.c:256]: Duble attempt to record-route Is it really so that record routing is not a branch specific thing? Is there a way to solve this problem? -- Juha

3 4

Kamailio+IMS+WebSocket
by mohammed alyaseen 26 Apr '14

26 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo Mohammed Al-khameri Technical University Berlin TUB

1 0

Re: [SR-Users] cnxcc
by Andrzej Ciupek 26 Apr '14

26 Apr '14

Hello I’am using default config with options from an example. It looks like event route isn’t execuded after hangup. Apr 26 16:26:14 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:15 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:16 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:17 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:18 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:19 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: Setting up money based credit control Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: Setting up channel based credit control Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: CNXCC ROUTE: 1000 has 2 call(s) kamailio.cfg … … # ----- prepaid call controll ----- #!define DLG_FLAG 28 #!define CC_FLAG 29 loadmodule "dialog.so" modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") loadmodule "cnxcc.so" modparam("cnxcc", "dlg_flag", CC_FLAG) modparam("cnxcc", "credit_check_period", 1) #check every 1 second … … … route { … … if (!allow_source_address("2") and !allow_source_address("1")){ route(CNXCC); route(PSTN); } … } route[CNXCC] { $var(client) = "1000"; xlog("L_INFO", "Setting up money based credit control"); $var(credit) = "10"; # 10$ of credit $var(cost_per_sec) = "1"; # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second # if only one call is established, that call should last 9 seconds. if (!cnxcc_set_max_credit("$var(client)", "$var(credit)", "$var(cost_per_sec)", "$var(i_pulse)", "$var(f_pulse)")) { xlog("Error setting up credit control"); return; } $var(count) = -1; if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { xlog("Error getting customer's channel count"); } xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)"); if ($var(retcode) < -1) { xlog("Too many channels for customer"); sl_send_reply(403, "Forbidden"); if (!cnxcc_terminate_all("$var(client)")) { xlog("Error terminating customer's calls"); } else { xlog("Terminating customer's call"); } exit; } } route[SHOW_CHANNEL_COUNT] { $var(count) = @cnxcc.channels["1000"].count; xlog("L_INFO", "1000 has $var(count) call(s)"); } event_route[dialog:failed] { xlog("dialog failed"); } event_route[cnxcc:call-shutdown] { xlog("L_INFO", "Shutting down channel by callcontroll"); xlog("L_INFO", "[$ci]: call killed"); $var(count) = $var(count)-1; if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { xlog("Error getting customer's channel count"); } xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)"); # perform some kind of notification, database update, email sending, etc } Greetings Andrzej Ciupek Od: Carlos Ruiz Díaz Wysłano: ‎sobota‎, ‎26‎ ‎kwietnia‎ ‎2014 ‎15‎:‎19 Do: Kamailio (SER) - Users Mailing List, Andrzej Ciupek Hi, sorry guys, I forgot about this. I will make my own tests and post the results. Andrzej, can you post the configuration file you are using? Regards, Carlos On Sat, Apr 26, 2014 at 4:15 AM, Kelvin Chua <kelchy(a)gmail.com> wrote: No, I have not yet found the solution for this issue on my setup. Maybe Carlos can help us out On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> wrote: Hello Have You solved this issue ? I’am using Kamailio 4.1.3 and I’am getting the same results. Greetings Andrzej Ciupek _______________________________________________ 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 -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

2 1

Re: [SR-Users] cnxcc
by Kelvin Chua 26 Apr '14

26 Apr '14

No, I have not yet found the solution for this issue on my setup. Maybe Carlos can help us out On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> wrote: > Hello > > Have You solved this issue ? > I’am using Kamailio 4.1.3 and I’am getting the same results. > > Greetings > Andrzej Ciupek >

2 1

Kamailio+IMS+WebSocket
by mohammed alyaseen 26 Apr '14

26 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo Mohammed Al-khameri Technical University Berlin TUB

1 0

LCR from_uri help
by Geoffrey Mina 26 Apr '14

26 Apr '14

Greetings, I need a little help with the LCR module. I am running Kamailio 4.1.2. The LCR module is behaving in a way that is inconsistent with our old 1.5 versions. We have a customer that passes a certain caller-id. That from_uri should be routing over a specific trunk. Here are the relevant bits of the LCR data from kamcmd. I would expect gw_index: 7 to match before gw_index: 6 here, but that is not the case. Any help would be greatly appreciated. Please note, i Have stripped out a bunch of other routes that are not in play here... hence the reason gw_index starts at "6". kamcmd> lcr.dump_gws { lcr_id: 1 gw_id: 11 gw_index: 6 gw_name: Route A scheme: sip ip_addr: 1.2.3.4 hostname: port: 5060 params: transport: UDP strip: 0 prefix: tag: flags: 2 defunct_until: 0 } { lcr_id: 1 gw_id: 1 gw_index: 7 gw_name: Route P scheme: sip ip_addr: 4.3.2.1 hostname: port: 5060 params: transport: UDP strip: 0 prefix: DLR# tag: flags: 0 defunct_until: 0 } kamcmd> lcr.dump_rules { lcr_id: 1 rule_id: 31 prefix: +1 from_uri: 2089328663 request_uri: stopper: 0 } { gw_index: 6 priority: 1 weight: 1 } { lcr_id: 1 rule_id: 5 prefix: +1 from_uri: request_uri: stopper: 0 } { gw_index: 7 priority: 1 weight: 1 }

2 7

Kam 4.1 Crash: HELP PLZ!!!!
by Samuel Ware 26 Apr '14

26 Apr '14

Here are the logs from the first crash Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7feac6836310,65572,0,205.251.172.14:5060,16): Message too long(90) Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7feaace02c98,65533,0,205.251.172.14:5060,16): Message too long(90) Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7feac6836310,65569,0,205.251.172.14:5060,16): Message too long(90) Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:32 tel-vc-fs03 /usr/local/sbin/kamailio[7589]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7feaca6c9738), called from <core>: mem/shm_mem.c: sh_realloc(88), first free <core>: mem/shm_mem.c: sh_realloc(88) - aborting Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: ALERT: <core> [main.c:775]: handle_sigs(): child process 7589 exited by a signal 6 Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7604]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7597]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7603]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7601]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Logs from second crash Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f34ec4f6ed8,65569,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f35242e5d28,65535,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f3516fb6720,65578,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f37d8e91088,65522,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: sl [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f37d8e91088,65522,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3961]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f37d81fa960,65522,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3961]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3974]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f34d7d6b720), called from <core>: mem/shm_mem.c: sh_realloc(88), first free <core>: mem/shm_mem.c: sh_realloc(88) - aborting Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3935]: ALERT: <core> [main.c:775]: handle_sigs(): child process 3974 exited by a signal 6 Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3935]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3935]: INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3985]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3983]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3966]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3988]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3972]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3984]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received

5 10

Kamailio+IMS+WebSocket
by mohammed alyaseen 25 Apr '14

25 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo

1 0

Re: [SR-Users] Minimalist Kamilio configuration example using dynamic routing module
by Gary Wallis 25 Apr '14

25 Apr '14

Hi, We are moving away from OpenSIPS and would like to start testing Kamailio. I really liked how Kamailio complies so clean and is configured easily in comparison with OpenSIPS monstrosity. So to not waste time can anybody provide some practical info for... Kamailio complilation and config examples (w/MySQL) to route Carrier SIP (DID incoming and outgoing PSTN termination) traffic to and from Asterisk PBXs. Example current OpenSIPS setup using dynamic routing module: 1-. dr_rules have the complete DID for PBXs (for incoming traffic from Carriers to be proxied to the correct PBX). 2-. dr_rules have the partial DID for Carrier gateways (for example based on internatinal, state, etc. routing of outgoing traffic from PBXs to Carriers). 3-. dr_gateways have the IP numbers for PBXs. 4-. dr_gateways have the IP numbers for Carriers. Any info and pointers appreciated. Best regards, Gary

2 2

Re: [SR-Users] Caller ID fields incorrect on Polycom 650s
by J Lee 25 Apr '14

25 Apr '14

Thanks for your response Daniel! I think I omitted a key piece of information in my original question! We are using the SCA module. Hope that sheds some more light on the situation. -jl > Date: Thu, 24 Apr 2014 23:06:28 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Caller ID fields incorrect on Polycom 650s > Message-ID: <53597CD4.8030106(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > kamailio is not touching from/to headers by default. If there are > incorrect, then could be somewhere else. You ahve tos end a sip trace > (ngrep for example) with incoming and outgoing sip packets on kamailio > server. > > Cheers, > Daniel > > On 24/04/14 21:58, J Lee wrote: >> Hi there. >> >> I have a situation where the caller id information on my phone appear >> incorrectly. >> When I have ext 1 dial ext 2, on the caller's phone, instead of saying >> "To: 2", it says "From: 2". >> >> I'm using kamailio version 4.0.4. I'm just new to everything kamailio >> and VoIP so I apologize in advance for any remedial questions. >> >> So far, I've done a SIP trace to and checked all packets line by line >> to see if I could spot where the To address was erroneously changed to >> FROM. In particular, I was checking the call-info fields for caller >> id data or any other data what would appear on the LED of the phone. >> >> Now I'm wondering, who's job (aka which module) is it to set the >> caller id information for both phones? >> I'm thinking of digging inside the code for the right module just to >> see if I can follow what's going on >> >> In addition to answering the question above, if you have any >> suggestions on what else I can check or try, I'd appreciate it. >> >> Thanks List! >> >> >> >> >> >> _______________________________________________ >> 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://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v4.1.3 Released
by Daniel-Constantin Mierla 25 Apr '14

25 Apr '14

Hello, Kamailio SIP Server v4.1.3 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.2. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.3. For more details about version 4.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/04/kamailio-v4-1-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Caller ID fields incorrect on Polycom 650s
by J Lee 24 Apr '14

24 Apr '14

Hi there. I have a situation where the caller id information on my phone appear incorrectly. When I have ext 1 dial ext 2, on the caller's phone, instead of saying "To: 2", it says "From: 2". I'm using kamailio version 4.0.4. I'm just new to everything kamailio and VoIP so I apologize in advance for any remedial questions. So far, I've done a SIP trace to and checked all packets line by line to see if I could spot where the To address was erroneously changed to FROM. In particular, I was checking the call-info fields for caller id data or any other data what would appear on the LED of the phone. Now I'm wondering, who's job (aka which module) is it to set the caller id information for both phones? I'm thinking of digging inside the code for the right module just to see if I can follow what's going on In addition to answering the question above, if you have any suggestions on what else I can check or try, I'd appreciate it. Thanks List!

2 1

Problem accessing git repository
by arun Jayaprakash 24 Apr '14

24 Apr '14

Is there a problem with the git repository? When I try to download kamailio (git clone --depth 1 git://git.sip-router.org/sip-router kamailio ) I get an error saying that git.sip-router.org can not be reached. Please let me know, thank you. Arun

2 2

Re: [SR-Users] TCP reset
by Ciprus, Daniel 24 Apr '14

24 Apr '14

Understand :-) thank you Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Carsten Bock Date:04/24/2014 6:09 AM (GMT-05:00) To: "Ciprus, Daniel" Cc: "Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] TCP reset Hi Daniel, Define "near future"... :-) Once we have time, we may look at this topic; but no timeline yet.... Kind regards, Carsten 2014-04-23 15:55 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>: > Thanks Carsten, > > Any plans to do so in near future ? :-) > > On 04/23/2014 09:52 AM, Carsten Bock wrote: > > Hi Daniel, > > what you want, is basically already implemented in the "Standard" > usrloc module, see: > > http://kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.handle_… > > We haven't implemented that feature yet in the IMS modules.... > > Schöne Grüße, > Carsten > > > 2014-04-23 15:47 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>: > > Hi, > > Is there any way to remove registration entry from registrar (Kamailio IMS > built on stable 4.1) detected on p-cscf by incoming TCP RST from the network > ? What's happening is that client is loosing connections/crashing and TCP > RST is sent back to pcscf which is not propagating changes back to > registrar. This is likely not possible with my version of kamailio but at > least logging which would indicate change of the status will be helpful. > > thanks for any hints > > > -- > Daniel Ciprus > Integration engineer > http://www.acision.com > > 9954 Mayland Dr > Suite 3100 > Richmond, VA 23233 > USA > T: +1 804 762 5601 > E: daniel.ciprus(a)acision.com > > ________________________________ > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you for > understanding. > > > _______________________________________________ > 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 Ciprus > Integration engineer > http://www.acision.com > > 9954 Mayland Dr > Suite 3100 > Richmond, VA 23233 > USA > T: +1 804 762 5601 > E: daniel.ciprus(a)acision.com > > ________________________________ > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you for > understanding. > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

TCP reset
by Daniel Ciprus 24 Apr '14

24 Apr '14

Hi, Is there any way to remove registration entry from registrar (Kamailio IMS built on stable 4.1) detected on p-cscf by incoming TCP RST from the network ? What's happening is that client is loosing connections/crashing and TCP RST is sent back to pcscf which is not propagating changes back to registrar. This is likely not possible with my version of kamailio but at least logging which would indicate change of the status will be helpful. thanks for any hints -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 3

Planning release of v4.1.3
by Daniel-Constantin Mierla 24 Apr '14

24 Apr '14

Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 8

Difference between route(RELAY) and t_relay
by aawaise 24 Apr '14

24 Apr '14

Hi all, I want to know while coding kamailio.cfg. What factor determines that route(RELAY) should be used or t_relay ?? Thanks. Cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Difference-between-route-RELAY-and-… Sent from the Users mailing list archive at Nabble.com.

2 1

LDAP and Kerberos backend for authentification / Or PAM / SASL ?
by Yoann Gini 23 Apr '14

23 Apr '14

Hello, My users are managed on a common LDAP / Kerberos setup. I don’t have access in any ways to the user password in clear text mode. I’ve start to follow this tutorial http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth and check this documenation http://kamailio.org/docs/modules/4.1.x/modules/ldap.html But it seems LDAP module provided in Kamailio don’t handle this scenario, it except to have user password in clear text mode in the LDAP database (wrong in so many ways…). Do we have a way with existing Kamailio plugin to use a authentication backend who don’t provide access to clear text password? I provide different authentication « adapter » who can help any service not compatible with Kerberos to handle authentication: - ldap bind - sasl - pam Do we’ve a way to use one of this backend as an authentication service (and not just a users database). Best regards, Yoann

2 4

IBM SIP server has implemented proxy managed overload protection
by Yang Hong 23 Apr '14

23 Apr '14

Hello Folks. According the new update on Feb 5, 2014, IBM has implemented proxy managed overload protection (PMOP) at SIP proxy server. http://pic.dhe.ibm.com/infocenter/wasinfo/v7r0/index.jsp?topic=%2Fcom.ibm.w… The above IBM link lists IBM SIP proxy server custom properties for SIP overload protection/control. •burstResetFactor •deflatorRatio •dropOverloadPackets •inDialogAveragingPeriod •maxThroughputFactor •outDialogAveragingPeriod •perSecondBurstFactor •proxyTransitionPeriod •sipProxyStartupDelay IBM SIP overload protection mechanism can be regarded as "Local SIP Overload Protection/Control" (Proceedings of WWIC, June 2013). I am moderator of open-source FreeRDP-WebConnect project at GitHub. https://github.com/FreeRDP/FreeRDP-WebConnect/issues/36 I will try to integrate two implicit SIP overload protection/control algorithms (RRRC, IEEE Globecom 2010 and RTDC, IEEE ICC 2011) into Kamailio (OpenSER) in the future. A survey on SIP overload protection/control algorithms (including IETF RFC "SIP Overload Control") can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, “A Comparative Study of SIP Overload Control Algorithms,"Network and Traffic Engineering in Emerging Distributed Computing Applications, Edited by J. Abawajy, M. Pathan, M. Rahman, A.K. Pathan, and M.M. Deris, IGI Global, 2012, pp. 1-20. http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SI… http://arxiv.org/abs/1210.1505 This survey paper provides a short review on SIP Express Router (SER). As we know, SIP Express Router (SER) and Kamailio (OpenSER) are open-source SIP router. "SIP Express Router (SER) provides a load balancing module to mitigate the overload caused by large subscriber populations or abnormal operational conditions (IP Telecommunications Portal, 2011). http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67… The presentation slides for two implicit SIP overload protection/control algorithms (RRRC and RTDC) are available for your download. Redundant Retransmission Ratio Control (RRRC) - implicit SIP overload protection/control algorithm (IEEE Globecom 2010 Slides) can be downloaded from the following ResearchGate link. https://www.researchgate.net/publication/258555827_Mitigating_SIP_Overload_… Round-Trip Delay Control (RTDC) - implicit SIP overload protection/control algorithm (IEEE ICC 2011 Slides) can be downloaded from the following ResearchGate link. https://www.researchgate.net/publication/257945199_Round-Trip_Delay_Control… The paper with Redundant Retransmission Ratio Control (RRRC, implicit SIP overload protection/control) algorithm can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, "Mitigating SIP Overload Using a Control-Theoretic Approach," Proceedings of IEEE Globecom, Miami, FL, U.S.A, December 2010. https://www.researchgate.net/publication/221284946_Miigating_SIP_Overload_U… http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5683124 Redundant Retransmission Ratio Control (RRRC, implicit SIP overload protection/control) algorithm has been quickly adopted by The Central Weather Bureau of Taiwan for their early earthquake warning system. Ting-Yun Chi, Chun-Hao Chen, Han-Chieh Chao, and Sy-Yen Kuo, "An Efficient Earthquake Early Warning Message Delivery Algorithm Using an in Time Control-Theoretic Approach", 2011. http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15# http://www.ipv6.org.tw/docu/elearning8_2011/1010004798p_3-7.pdf Short review and comments on RRRC implicit SIP overload protection/control algorithm by former IEEE TAC Associate Editor S. Mascolo: Luca De Cicco, Giuseppe Cofano, and Saverio Mascolo, "Local SIP Overload Control", Proceedings of WWIC, June 2013. http://link.springer.com/chapter/10.1007%2F978-3-642-38401-1_16# http://c3lab.poliba.it/images/2/2a/SipOverload_WWIC13.pdf The paper with Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) algorithm can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, "Design Of A PI Rate Controller For Mitigating SIP Overload," Proceedings of IEEE ICC, Kyoto, Japan, June 2011. https://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Cont… http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5963029 Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) algorithm has been recommended as White Paper by TechRepublic (CBS Interactive) http://www.techrepublic.com/whitepapers/design-of-a-pi-rate-controller-for-… Control theoretic approaches have been applied to model the interactions between an overloaded SIP server and its upstream servers as a feedback control system in two different scenarios - Round-Trip Delay Control (IEEE ICC 2011) and Redundant Retransmission Ratio Control (IEEE Globecom 2010). Journal paper (implicit SIP Overload Protection/Control) published by Springer Telecommunication Systems not only conducts more theoretical analysis of Round-Trip Delay Control (RTDC) and Redundant Retransmission Ratio Control (RRRC), but also discusses how to apply RTDC algorithm to mitigate SIP overload for both SIP over UDP and SIP over TCP (with TLS). Journal paper can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, "Applying control theoretic approach to mitigate SIP overload", Telecommunication Systems, Volume 54, Issue 4, December 2013, pp 387-404. https://www.researchgate.net/publication/257667871_Applying_control_theoret… http://link.springer.com/article/10.1007/s11235-013-9744-8 Implicit SIP overload protection/control solution using control theoretic approaches can be regarded as an engineering applications of control theory, see the discussion on control system design in the answers to the following two ResearchGate(RG) questions. ResearchGate is the largest social network for research scientists and engineers in the world. RG Question #1: "What are trends in control theory and its applications in physical systems (from a research point of view)?" https://www.researchgate.net/post/What_are_trends_in_control_theory_and_its… RG Question #2: "What are the latest problems to be solved in "control of nonlinear systems"?" https://www.researchgate.net/post/What_are_the_latest_problems_to_be_solved… RG Question #3: "What are the new trends in control engineering?" https://www.researchgate.net/post/What_are_the_new_trends_in_control_engine… Best regards, Winston Hong Software Engineer Ottawa, Ontario Canada

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rtpproxy_manage() issues in rtpproxy-ng
by Alex Balashov 23 Apr '14

23 Apr '14

Hello, I'm running the latest pull of Kamailio 4.1 (last commit be187e135b0b9b28136817c3569ab5c0abcc5b3f) and am using rtpproxy-ng with a recent mediaproxy-ng master (commit cb6990e43864b077dd6a24acfbdf5ef76c1a427e). For no apparent reason, Kamailio has stopped sending 'offer' commands to it when I use rtpproxy_manage(). My use of it is in this setting: if(isflagset(PROXY_MEDIA) && !isflagset(PROXY_MEDIA_SET) && has_body("application/sdp")) { set_rtp_proxy_set("1"); rtpproxy_manage("o"); add_rr_param(";proxy_media=yes"); setflag(PROXY_MEDIA_SET); } I put in a log message to confirm that this block is being run. It is. But, there's no offer command going to the mediaproxy-ng on the other box, as confirmed by packet captures and logs. However, the mediaproxy-ng is getting lots of 'answer' and 'delete' commands that it doesn't know what to do with, since the call was never initialised with 'offer'. When I change rtpproxy_manage() to rtpproxy_offer(), it works perfectly. I've been using rtpproxy_manage() forever, almost as long as it's existed. Does it not work with rtpproxy-ng, despite what the documentation says? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 8

RTPProxy and premature RTP
by Daniel Tryba 23 Apr '14

23 Apr '14

I'm having some troubles with a provider sending RTP before a 183 Session Progress or 200 OK (I see up to 1s of rtp prematurely). The machine is running rtpproxy and apparently rtpproxy buffers these rtp packets and flushed them in one burst when the 183/200 arrive, this creates havock in some endpoints but is undesirable in all cases IMHO if the delay of the 182/200 is to high (>0.1s). Is there a way to control buffering/flushing from kamailio? Is rebuilding rtpproxy with a smaller buffer an option? Or should I switch to an other proxy module? Any thoughts about this subject (apart from getting the provider to stop sending premature rtp)? -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 2

Kamailio-Asterisk - route "FROMASTERISK" not found
by Patrik Kristel 23 Apr '14

23 Apr '14

Hello all, I'm trying to implement Kamailio 4.1 with Asterisk 12.1.0 regarging this tutorial: > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And when I try to compile kamaili.cfg, I still got this error: ERROR route "FROMASTERISK" not found I have loaded all modules like in tutorial. I tried to find some solve this issue, but with no result.

2 2

Re: [SR-Users] Drouting
by Keith 23 Apr '14

23 Apr '14

Hi, Or is it possible to use a regular expression for the user? I basically want certain users to use a certain route without having to add them all in. Keith

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