- sr-users - kamailio.org

Need a commercial SIP Service
by Karl Grimm 22 May '14

22 May '14

Dear iptel, Thankyou for your open source website and excellent information. I hope you are able to help me with the following. We are an Australian organisation. We are trying to find a good quality SIP provider, commercial grade, with high connectivity success rates. We have used Skype, and it works incredibly well. It gets through your firewall and succeeds with login and successful voice and video call connection just about every time. But every public or commercial SIP Service I have used has failed to perform as well. Both the open source and the commercial SIP servers all suffer from the same problems: a.. they don't login unless you open up port 5060 on your firewall, regardless of whether STUN or TURN or ICE is in use b.. they don't reliably connect each call, sometimes the calling end just keeps ringing, and the receiving end doesn't ring c.. even if a call gets connected, they often have one of the 4 A/V streams missing, eg. we get video and outgoing audio, but no incoming audio d.. when one party hangs up, sometimes the other end sometimes doesn't know about it and just sits there thinking the call is still underway e.. when the Internet is busy, eg. at 4pm when children get home from school and jump on the Internet, their media relays struggle to keep up and UDP streams are badly degraded Note that we have not yet tried iptel.org, but I note that video is not talked about much on your website, and also you are an open source organisation, who cannot be expected to provide a commercial grade service. (If I am wrong about this, please tell me and I will certainly give iptel.org a try!) Using Jitsi, Linphone, and many other softphones, we have tried .... a.. ippi.fr b.. jumblo.com c.. pennytel.com d.. opensips.org e.. sip.linphone.org f.. an Australian commercial VoIP service by MyNetFone (works very well, but voice only) We are a business, and we are prepared to pay for a SIP Service. We cannot use ooVoo, Tango, Goober, ippi Messenger and other closed applications. We have to use an open source softphone such as Jitsi or Linphone, because they have an API. So my questions are .... a.. Are you aware of any commercial grade SIP providers that work as well as Skype: a.. they have STUN / TURN / ICE support in place b.. they support both video and audio calls c.. login occurs reliably: a.. regardless of where I am located (at home, at work, on my smartphone, using WiFi at a hotel, etc.) b.. regardless of what firewall I am behind d.. call connection occurs reliably, regardless of where I am located and regardless of firewall a.. every call always connects b.. if codecs at each end match, the call successfully connects audio and video streams both ways c.. when one party hangs up, the other end knows about it and doesn't just sit there thinking the call is still underway e.. their servers and infrastructure performs well even at peak periods, for example at 4pm when all the children get home from school and jump on the Internet f.. they have high performance media relays in place, so that audio and/or video can be relayed if required b.. If no company or organisation offers a SIP Service that performs as well as Skype, can you recommend any company or individual that we can engage (pay) to establish our own SIP Service. They must have the skills to setup SIP Server, STUN Server, TURN Server, and to adjust ports and settings so that anybody can login and make a call from anywhere, behind any firewall. Your recommendations would be very much appreciated. Thankyou, Karl Grimm. -------------------------------------------------------------------------------- Karl Grimm, Managing Director Mobile: 0418 310 989 | karl(a)konnekt.net.au Konnekt feel connected, feel safe

2 1

avpops issue
by Gilbert T. Gutierrez, Jr. 22 May '14

22 May '14

I am trying to setup call forwarding but I am getting a failure when I attempt to implement it. Kamailio Version 4.1.1 Centos 6 x64 Using precompiled RPMs from telephony.repo I am following guidance from the following urls... http://www.kamailio.org/wiki/tutorials/mini-howto-admin/call_forwarding http://www.kamailio.org/dokuwiki/doku.php/tutorials:avpops http://www.kamailio.org/dokuwiki/doku.php/examples:set-blind-call-forwarding 0(16018) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command is_user_in 0(16018) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 1046, column 41: unknown command, missing loadmodule? 0(16018) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command avp_write 0(16018) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 1048, column 39: unknown command, missing loadmodule? My understanding is that when I load module avpops that it should include those 2 procedures (avp_write and is_user_in). I am loading the module avpops.so Can someone please set me straight. Thank you. Gilbert T. Gutierrez, Jr. Phoenix Internet

2 2

My kamailio.cfg has ofen not registred , please help me check my registrar process is right or wrong
by vast 22 May '14

22 May '14

Hi, My kamailio has ofen not registred ,please help me check my registrar process is right or wrong (I think the MSILO modle maybe tedious or incorrect).this is my kamailio.cfg file.

2 1

Kamailio with Sqlite
by Ashwin Kumar R 22 May '14

22 May '14

Hi, I want to configure kamailio to use SQLITE instead of MYSQL.I made DB_ENGINE=SQLITE in kamctlrc file and loadmodule "db_sqlite.so" in kamctl.conf file. But when I changed these parameters modparam("auth_db", "db_url", "sqlite:///etc/kamailio/kamailio.db") #!define DBURL "sqlite:///etc/kamailio/kamailio.db" Im getting the following error ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I checked the log file ,following is the comment in syslog May 20 19:16:30 r13pc80-desktop /usr/local/sbin/kamailio[10920]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'etc/kamailio/kamailio.db' What kamailio.cfg and kamctlrc parameters do I need to change to do this? Regards Ashwin

3 4

Accounting based on location/contact
by John Murray 22 May '14

22 May '14

Hello, I have subscribers that have 2 devices in the location table. These can make and receive call correctly and I can account for the calls. However I need to account which device has made or received the call as there are charges for one device type but not the other. I can modify the contact string from the device to add a parameter for example but I can't see how to add that into the accounting records using tm and acc modules. Thanks for your help. Regards John

3 3

i use the csipsimple registrar kamailio get request time out
by vast 22 May '14

22 May '14

Hi, I registrar kamailio ofen can registrar .but at other wifi some time i use the csipsimple registrar kamailio get request time out . when i look up the log I have found ERROR: registrar [save.c:692]: update_contacts(): invalid cseq for aor <1008> I search on google but can`t find a good answer. some one tell me it's not a same callid and low cseq . what would i do for this error ?

2 3

Asterisk Integration w Kamailio and Zoiper phone problems
by Brian Watson 22 May '14

22 May '14

Hi, I have a virtual environment setup and am trying to get the integration of Kamailio and Asterisk to a least be able to make a phone call based off the Asterisk Hello World example. I do the following: 1. I'm running Ubuntu 14.04 and Oracle Virtual Box 2. I Integrate Kamailio and Asterisk based off of the asipto kb: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb with both of them running on the same VM. 3. I've modified sip.conf and extensions.conf to have entries that tie together phones 101 and 102 3. I start asterisk with "asterisk" and "asterisk -rvvvvvv" 4. I install and setup the Zoiper Softphone as indicated in the tutorial. https://wiki.asterisk.org/wiki/display/AST/Hello+World 5. In Zoiper I hardcode the Domain: xxx.xxx.xxx.xxx:5080 and also enter the username and password. 6. I'm doing all of this without access to an internet connection so that's why I hardcode the IP address. 7. After setting of the Zoiper softphone I try to get it to communicate with Asterisk, but I get a 408 request timeout. Any ideas as to what might be wrong? Thanks, Brian

3 5

Kamailio and RTPProxy
by Thyrus Gorges 22 May '14

22 May '14

Hello all, I am currently trying to configure Kamailio and rtpproxy to play back a short message to the user-agent. I’ve started rtpproxy using: rtpproxy -F -s udp:127.0.0.1:7722 -l 10.1.10.102 -d WARN:LOG_LOCAL0 -u root I created the necessary sound file using makeann from the source of rtpproxy. Below are the relevant pieces of kamailio.cfg # ----- rtpproxy parameters ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") Kamailio.cfg ————— # Validate that the User Agent via IP Authorization via a SQL table if (!allow_address("1", "$si", "$sp")) { sl_send_reply("403", "Forbidden"); xlog("**** $si is UNAUTHORIZIED ****"); exit; } xlog("Enter route 1"); if (has_body("application/sdp")) rtpproxy_answer(); if (is_method("INVITE")) { xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]"); if (rtpproxy_offer()) { t_on_reply("1"); } else { t_on_reply("2"); } #t_on_branch("2"); #t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[1] { if (has_body("application/sdp")) rtpproxy_manage(); rtpproxy_stream2uas("/var/rtpproxy/prompts/connect", "-1"); exit; } onreply_route[2] { if (has_body("application/sdp")) rtpproxy_offer(); exit; } When starting Kamailio, it informs me that it found the proxy and it enabled support for it. May 21 20:44:50 localhost /usr/local/sbin/kamailio[23816]: INFO: rtpproxy [rtpproxy.c:1607]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled Now, once I start up a call, rtpproxy reports the following information: May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: new session MTFiNWU1N2NiZGQwNWMwYzVkN2MzZjUyNzhkNTU3YWE., tag 88fc4d50;1 requested, type strong May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: new session on a port 41236 created, tag 88fc4d50;1 May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: pre-filling caller's address with 10.8.0.10:8000 May 21 20:14:35 localhost /usr/local/sbin/kamailio[23528]: ERROR: <script>: rtpproxy_offer returned true May 21 20:14:35 localhost /usr/local/sbin/kamailio[23522]: ERROR: rtpproxy [rtpproxy.c:1771]: select_rtpp_node(): script error -no valid set selected May 21 20:14:35 localhost /usr/local/sbin/kamailio[23522]: ERROR: rtpproxy [rtpproxy_stream.c:113]: rtpproxy_stream(): no available proxies Any assistance is greatly appreciated ! — Thyrus Gorges

1 0

Presence
by Joel White 21 May '14

21 May '14

I am working on enabling presence and bla in a Kamailio + FreeSWITCH environment All handsets are Polycom with a handful of Grandstream ATA's I enabled presence and presence_xml I have not got it to work as I keep seeing this meesage in the logs NOTICE: presence [subscribe.c:1030]: handle_subscribe(): Unsupported presence event call-info When I searched online for this error, the answers I got were saying this was a Linksys/Snom message.... How do I get presence to work properly? Thank you in advance

4 5

uri parameter value case sensitiveness
by Daniel-Constantin Mierla 21 May '14

21 May '14

Hello, hopefully someone here with the RFC fresh in mind and its grammar for sip uri parameter can jump on to clarify if the value of the parameter can be changed to lower/upper case. For a specific example, is it allowed that the value of a parameter in contact header uri is changed to upper/lower case when building the sequential request (so this appears in request uri)? Same would be for Recor-Route uri parameter that would appear later in Route headers... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 8

Realtime integration: Unregistered clients showing as registered?
by Olli Heiskanen 20 May '14

20 May '14

Hello, One of the tests I've been working with is Asterisk realtime integration according to Daniel's guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb Weird thing is the client looks registered but I'm not sure if it really is registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this: *CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] Also, calling between clients will fail; in Asterisk cli I get: *CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: Dialed 661") in new stack -- Executing [661@default:2] Dial("SIP/660-00000000", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/661 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup("SIP/660-00000000", "") in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000000' In this setup I have SIP peers in an asterisk table added like this: INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com '); I have Kamailio and Asterisk on the same machine where Kamailio listens port 5060 and Asterisk listens 5070. Things that differ from the guide are Kamailio and Asterisk versions, which in my case are newer. Also, for another testing case I have MULTIDOMAIN enabled in Kamailio, does this interfere with the realtime integration? I'm using only one domain though. Please let me know if any configs or traces I can provide will help figure out what's going on. cheers, Olli

4 10

Upgrade to version 4.1
by VOIP Tests 20 May '14

20 May '14

Hello, I understand that there are some bugs in ver 4.03. Can someone let me know how I can update my running version of Kamailio ( 4.03 ) to version 4.1? Will I need to do a fresh install of the new version or is there a way I can do an upgrade/update to ver 4.03? Thank you, Arun

2 2

[rtpengine] No media from WebRTC UA
by Alexey Rybalko 20 May '14

20 May '14

Hello! During a call from classical SIP softphone to WebRTC there's no media from the browser (Mozilla, the same result is for Chrome). In case of a call from the browser to the softphone there's media flow from both sides. The snippets from kamailio.cfg related to the problem case (SIP-->WebRTC) are below. OFFER: $var(rtpp_flags) = "trust-address symmetric replace-origin replace-session-connection"; $var(rtpp_flags) = $var(rtpp_flags) + " ICE=force"; $var(rtpp_flags) = $var(rtpp_flags) + " RTP/SAVPF"; rtpengine_offer($var(rtpp_flags)); ANSWER: $var(rtpp_flags) = "trust-address symmetric replace-origin replace-session-connection"; $var(rtpp_flags) = $var(rtpp_flags) + " ICE=remove"; $var(rtpp_flags) = $var(rtpp_flags) + " RTP/AVP"; rtp.log is attached. Any help on this issue would be very appreciated. with best regards, Alexey

4 6

code review tools
by Daniel-Constantin Mierla 20 May '14

20 May '14

Hello, some time ago it was proposed to use a code review tool to analyze patches, the topic was also discussed during the last IRC devel meeting. I cc-ed users mailing list, because people there might have experiences with some of them. I found on wikipedia a short list of available applications for this purpose: - http://en.wikipedia.org/wiki/List_of_tools_for_code_review Personally I haven't used any so far. Iirc, gerrit was proposed before by Victor Seva. From the wikipedia list, Differential would be straightforward to deploy from dependencies point of view, as we have php on our servers. But I have no clue which is the right/best to use, so here we have to decide if worth to use one and which one. Besides the above one, perhaps github has an integrated review tool. I can be considered as an alternative as well. My opinion is that such tool is useful, specially for big patches, to allow easier more people to have a look at them. Hoping to get feedback on benefits of such tools and which would be the right/best to use. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

failed to find command redis_cmd
by Ryan Brindley 20 May '14

20 May '14

Hey Community, I'm trying to play around with ndb_redis, but seem to be missing something probably obvious as I'm getting a "failed to find command redis_cmd" error on load. loadmodule "ndb_redis.so" modparam("ndb_redis", "server", "name=test;addr=127.0.0.1;port=7000") request_route { redis_cmd("test", "SET foo bar", "r"); } I've verified that mpath = /usr/local/lib64/kamailio/modules/ and that ndb_redis.so is in that directory. Any help is greatly appreciated. Ryan Brindley

3 9

cfg values read from Database possible?
by Daniel Vukicevic 19 May '14

19 May '14

kamailio -V version: kamailio 3.2.2 (x86_64/linux) I am setting different cfg Parameters via XMLRPC. Which is good for changing parameters per runtime and being independent from database lookups when kamailio is running. Unfortunately cfg settings will be lost and empty after Restarting Kamailio if cfg values aren’t set before the request_route. So the idea was to look if cfg value is set, if not look into database and take the last cfg setting which was set. (Last cfg values will updated via SQL Query in the XMLRPC branch). # check if abc.test is defined if (strempty($sel(cfg_get.abc.test))) { sql_xquery("ca", "SELECT * from cfg where abc=‚test'", "ra“); The problem is that it seems, that there is no way to read out the values from database and to set the cfg variable again when kamailio is starting, because there is no way to inject cfg values into kamailio, when kamailio is already started. Here i need something like this: abc.test = "$xavp(ra=>LB01)" desc „TEST“; <<which doesn’t work in this form. (just to make it clear what i want) I found http://kamailio.org/docs/modules/stable/modules/cfg_db.html , but documentation is quite bad. Are there any other possibilities for me? The only way it might work for me (which is in my opinion quite dirty), to exec_dset an external command and to inject the new cfg via XMLRPC. The new cfg value is updated into database via XMLRPC again (which works, just for the sake of completeness): #!ifdef WITH_XMLRPC route[XMLRPC] { … … # Keeping values for Kamailio Restart sql_query("ca", "UPDATE cfg set LB01='$sel(cfg_get.abc.test)' where abc=‚test‘"); Mabye of one of you guys have a hint for me to solve this problem? Cheers Daniel

2 4

kamailio-OpenIms Presence
by Ashwin Kumar R 19 May '14

19 May '14

Hi, At the moment I'm working in a project, in which I need to use OpenIMSCore with a presence server, so I decided to use Kamailio.What should be added in the kamctlrc and kamailio.config files to make it meet the above requirement. Regards Ashwin

1 0

can you give me the complete kamailio.cfg that can use rtpproxy and uac_redirect?
by vast 19 May '14

19 May '14

Hello, lists. I was kamailio beginners, it seems demo on uac_redirect modules, but does not know how to use the kamailio.cfg file, can you give me an example of a complete hair. Thank you

1 0

Rtpproxy ptime in ringing inject
by Grygoriy Dobrovolskyy 19 May '14

19 May '14

Hello list, a long time ago i have build and application which injects ringing from rtpproxy to the client, while i am trying to reach his destination. The commande looks like this (log) May 19 10:17:44 rtpproxy[2529]: DBUG:handle_command: received command "29304_278705 P-1 B2B.256.470174 /usr/local/src/rtpproxy/sounds/ring session d20fbd749deed7e1f54460af276dee6d-0ead;1 " May 19 10:17:44 rtpproxy[2529]: INFO:handle_play: -1 times playing prompt /usr/local/src/rtpproxy/sounds/ring codec 8 May 19 10:17:44 rtpproxy[2529]: DBUG:doreply: sending reply "29304_278705 0#012" Everything works fine, a long time ago i have modded the rtpproxy a bit so when he sends the audio, he sends it with 20ms ptime instead of default 10ms, and i have totally forgot where is that part, i am digging into sources and not able to find it. Dear list if somebody remembers where is this damn setting in sources, i would be glad to hear from you. Thank you

1 0

mtree reload commands don't remove items from memory
by Juha Heinanen 18 May '14

18 May '14

i started to wonder why routing was sometimes wrong in my proxy and after studying the issue, i came to the conclusion that mtree reload commands (mi or rpc) don't remove entries from memory when they are removed from database. i started my proxy with empty mtrees db table (and empty memory table): # sip-proxy_ctl mi mt_summary MT:: TNAME=lp_out TTYPE=0 MEMSIZE=0 NRNODES=0 NRITEMS=0 RELOADCOUNT=0 RELOADTIME=1400419785 then i added one entry to db table and gave reload command. after that one entry correctly showed up in memory table: # sip-proxy_ctl mi mt_summary MT:: TNAME=lp_out TTYPE=0 MEMSIZE=1636 NRNODES=6 NRITEMS=1 RELOADCOUNT=1 RELOADTIME=1400419817 then i deleted the only entry from db, verified with phpmyadmin that the table really was empty, and gave again reload command. that command didn't have any effect in memory table: # sip-proxy_ctl mi mt_summary MT:: TNAME=lp_out TTYPE=0 MEMSIZE=1636 NRNODES=6 NRITEMS=1 RELOADCOUNT=1 RELOADTIME=1400419817 not even reload count got incremented. after that i went and added some debug to rpc_mtree_reload function and got this: May 18 16:30:17 siika /usr/sbin/sip-proxy[26536]: INFO: mtree [mtree_mod.c:1169]: rpc_mtree_reload(): reloading from db 'lp_out' the info is here: LM_INFO("reloading from db '%.*s'\n", pt->tname.len, pt->tname.s); if(mt_load_db(pt)!=0) { LM_ERR("cannot re-load mtree from database\n"); goto error; } my conclusion is that mt_load_db() function is somehow broken. -- juha

2 6

Siremis how to
by Joli Martinez 18 May '14

18 May '14

Hello, I am looking for some documentation or a how to on setting up some carrier routes using Siremis. thanks,

1 0

gateway connection
by Joli Martinez 18 May '14

18 May '14

Hello, I have setup Kamailio with Freeswitch and it is working. Right now I have my SIP providers connected to Freeswitch. Would it make sense to connect my SIP providers to kamailio instead. My setup will be intended to resell minutes. If so what advantages will I gain by connecting my SIP providers to kamailio

2 2

fetch bindings- timeout
by Daniel Ciprus 16 May '14

16 May '14

All, According to RFC3261: 10.2.3<http://tools.ietf.org/html/rfc3261#section-10.2.3> Fetching Bindings A success response to any REGISTER request contains the complete list of existing bindings, regardless of whether the request contained a Contact header field. If no Contact header field is present in a REGISTER request, the list of bindings is left unchanged. We should be able to fetch bindings from registrar. With my very limited knowledge of kamailio code (dialog-ng/dlg-handlers.c) : if ((!msg->cseq && (parse_headers(msg, HDR_CSEQ_F, 0) < 0 || !msg->cseq)) || !msg->cseq->parsed) { LM_ERR("bad sip message or missing CSeq hdr :-/\n"); goto error0; } This condition should also have something like: msg->method != "REGISTER" otherwise it's throwing errors and I'm not getting any response back from kamailio. Request just times out. Can somebody confirm that this feature is/is not implemented ? thank you Dan. Here is snippet of sipp xml scenario which I'm using to fetch bindings: <scenario name="FetchBindings"> <label id="1"/> <send> <![CDATA[ REGISTER sip:[field2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch] From: <sip:[field0]@[field2]>;tag=aabbcc To: <sip:[field0]@[field2]> Call-ID: [call_id] CSeq: [cseq] REGISTER Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER User-Agent: IM-client/OMA1.0 sipp-script P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 Privacy: none P-Visited-Network-ID: [field2] Supported: path Content-Length: 0 ]]> </send> <recv response="200" timeout="10000"/> </scenario> -- 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 1

Just 80 call/second
by Andrea Meroni 16 May '14

16 May '14

Hi all, i have some strange performance issues (with Kamailio config file i guess): using the default configuration file i can reach a 500 call/s rate, while using the default configuration file in the dispatcher folder i can reach just a 80 call/s rate (using the dispatching failover algorithm ...). The dispatcher is in front on a very basic Kamailio-based IMS core. Something has some hint about this issues? Andrea

2 4

Dispatcher poor performance
by Andrea Meroni 16 May '14

16 May '14

Hi all, i have a strange performance problem: i placed a dispatcher in front of a bundle of Kamailio proxies and, even with a really simple dispatching logic route { #ds_select_dstl("1", "2"); $du="sip:192.168.100.51:5060"; forward(); } that actually doesn't even use the dispatcher module i get that when the server's (192.168.100.51) CPU is 100% used the dispatcher's CPU is ~50% used. Using the dispatching algorithm based on the hash of the From-URI or the failover one (that should be the lightest dispatching algorithm, computationally speaking ...) the performance are almost the same. So, since i saw that in the documentation that the dispatcher module is designed to be "suitable for handling heavy SIP traffic", my question is: what i misconfigured? In my opinion it seems weird that the dispatcher module is able to manage just (as my tests suggest) just 2 Kamailio SIP proxies ... Andrea

1 0

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