sr-users April 2010

sr-users@lists.kamailio.org

65 participants
103 discussions

[Fwd: Re: [Kamailio-Users] Possible timing/latency relatedproblem ?]
by Jeff Brower 19 Apr '10

19 Apr '10

Juha- Vikram had posted a while back on the thread where he explained a bit more about his resolution and the fact the issue was occurring with VoipSwitch but not asterisk. Let us know if you want a link to that. Thanks for following up. -Jeff -------------------- Original Message ----------------- Subject: Re: [SR-Users] [Kamailio-Users] Possible timing/latency relatedproblem ? From: "Vikram Ragukumar" <vragukumar(a)signalogic.com> Date: Mon, April 19, 2010 11:15 am To: "Juha Heinanen" <jh(a)tutpro.com> Cc: users(a)lists.kamailio.org ------------------------------------------------------- Juha, > what was the conclusion regarding this? did the problem go > away when you called nat traversal functions both on 180 > ringing and 183 session progress? Yes. The problem did go away once we started forcing rtpproxy on 180 Ringing also. Thanks and Regards, Vikram.

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[Kamailio-Users] Possible timing/latency related problem ?
by Vikram Ragukumar 19 Apr '10

19 Apr '10

Hello, Scenario 1 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|--|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Lowlatency->| Scenario 2 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|---|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Highlatency->| Both scenario 1 and scenario 2 are similar in setup, excepting that in scenario 2 there is a greater latency involved in SIP and RTP transmission between the wireless router and Kamailio+rtpproxy server. In scenario 1 calls from the cellphone using a SIP softphone app go through 100% of the time with both endpoints of the call being audible. However in scenario 2, calls from the cellphone go through with both endpoints of the call being audible only sometimes, and during all other attempts, call goes through with no voice from the cellphone. In scenario 2 the SIP softphone can always register, the SDP connection address is always rewritten to use rtpproxy so it does not appear to be a NAT issue, rtpproxy's logs show a session being set up and torn down, with RTP stats for caller being 0. In both scenarios, the equipment and their configuration are identical. (Both cellphone's are Nokia e51, router's are Netgear wgr614) What might be probable causes for the behavior seen in scenario 2 ? Could there be a timing/latency related issue ? Any pointers on how to approach this problem will be appreciated. Thanks and Regards, Vikram.

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loadbalancer
by Hector Muñoz 19 Apr '10

19 Apr '10

Hi all, I have a kamailio sip server with two trunk connections to two asterisk servers. The register server is kamailio. I'd like to be able to configure a load balancing with kamailio between two asterisk servers. How can i do that? Do you know any good tutorial for configure load balancing with dispatcher module in kamalio? Thanks in advance. Regards. -- jabber: triptik(a)12jabber.com blog: http://impresionesdeunloco.wordpress.com

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sr-trunk with mysql support
by Meftah Tayeb 17 Apr '10

17 Apr '10

hi dear all, i'm trying to use latest trunk with mysql i have down compiling kamailio now i don't know ghow do i enable mysql for kamailio (not for kamdbctl) the db is created successfully but kamailio is unable to connect to the db default user/pass, i don't change anything how do i use this define stufs? thanks

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load balancing redirect
by Benjamin Lawetz 17 Apr '10

17 Apr '10

Hi, I'm trying to setup Kamailio to act as a load-balancer in front of asterisk servers. I want it to do the following: - The UA sends an invite to Kamailio - Kamailio sends a 30x redirect to the UA (dispatcher) - The UA sends it's invite to the asterisk server I've seen a couple of examples, but doesn't seem to work for me. Here's the relevant config: route{ if (method=="REGISTER") { [...clip...] }; if (method=="INVITE") { ds_select_domain("2","8"); #ds_select_dst("2","8"); sl_send_reply("300","Redirect"); exit; }; } What seems to happen is that Kamailio does send the redirect, but with no indication of where to go. ds_select_domain (or ds_select_dst ... I having trouble understanding the difference like many people it seems) I'm guessing selects an appropriate uri, but doesn't put it the contact for the redirect. Is there anyway to see what was set by the ds_select_dst? what pv would have this value? What do I need to set prior to sending a redirect reply for it to be properly forwarded? BTW: using Kamailio 3.0.1 Thank you, -- Benjamin

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wiki: upgrade from 3.0.x to 3.1.x
by Daniel-Constantin Mierla 17 Apr '10

17 Apr '10

Hello, I started the page that should collect notes about how to upgrade from version 3.0.x to 3.1.0: http://sip-router.org/wiki/install/3.0.x-to-3.1.x 3.1.0 is not yet released, target date is September 2010, right now is the code on master branch in GIT repository. However, this page will help people that want to test it or use some modules from it, making as well the life easier before release. Please add to the document as you change (or discover) something breaking compatibility with 3.0.x - the wiki is open. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

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reply when drop() in branch route
by jh＠tutpro.com 17 Apr '10

17 Apr '10

i call drop() in branch route and it is the only branch left. the branch gets correctly dropped, i.e., the request is not send out, but the reply to UAC is strange: U 2010/04/14 21:03:43.750712 192.98.102.10:5060 -> 192.98.103.2:5074 SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM). Via: SIP/2.0/UDP 192.98.103.2:5074;rport=5074;branch=z9hG4bKaoxzewgj. To: <sip:0407058055@192.98.102.10>;tag=c02670ad1171fe45d9ff9a27d6c2cb82-e03b. From: <sip:+35816234567@foo.bar>;tag=bclsw. Call-ID: doibilvynrtqxpk@localhost. CSeq: 134 INVITE. Server: SIP Proxy (3.0.99-dev1 (i386/linux)). Content-Length: 0. why this kind of reply, because the proxy didn't even try to send the request to next ho? is it possible to somehow choose in the script, which reply to send? in this example, i'm using lcr to choose gws and none of them was suitable, so something like '503 service not available' would be more appropriate. -- juha

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tm response issue
by Klaus Darilion 16 Apr '10

16 Apr '10

Hi! Kamailio does not forward the received 503 response but instead generates a new 500 response. Failure routes are not activated. U 2010/04/14 15:08:26.201865 83.136.22.111:5060 -> 83.136.32.159:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK2a5a.0d5ea987.0;received=83.136.32.159 Via: SIP/2.0/UDP 10.10.0.51:5086;received=83.136.33.3;rport=5086;branch=z9hG4bK2901533893 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=as1b74528e Call-ID: 3387977514 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 U 2010/04/14 15:08:26.201865 83.136.32.159:5060 -> 83.136.22.111:5060 ACK sip:01888801505641636@labs.nic.at SIP/2.0 Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK2a5a.0d5ea987.0 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=as1b74528e Call-ID: 3387977514 CSeq: 21 ACK Max-Forwards: 69 Content-Length: 0 # U 2010/04/14 15:08:26.201865 83.136.32.159:5060 -> 83.136.33.3:5086 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 10.10.0.51:5086;rport=5086;branch=z9hG4bK2901533893;received=83.136.33.3 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=c3829fdbb3bfc7416671e56cae8e633d-a9b5 Call-ID: 3387977514 CSeq: 21 INVITE X-Info: (83.136.32.159:1035) Enforcing NAT traversal for local caller X-Info: (83.136.32.159:1035) Enforcing symmetric response routing Server: kamailio (3.0.1 (i386/linux)) Content-Length: 0 Is this a feature or a bug? thanks klaus

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Re: [SR-Users] load balancing redirect
by Iñaki Baz Castillo 16 Apr '10

16 Apr '10

2010/4/16 Benjamin Lawetz <benjaminlawetz(a)gmail.com>: >> I strongly suggest not to do that. Instead: >> >> - The UA sends an invite to Kamailio >> - Kamailio uses LCR module to randomly select asterik-1 or asterisk-2. >> - Kamailio sends the request to the selected asterisk. >> - If there is a failure (to inspect in failure_route) Kamailio uses >> next_ge() (LCR module) and routes the INVITE to the other asterisk. >> >> This will work really better. > > Good idea, hadn't thought of a retry for the invite (was thinking of > removing the server from the dispatch if it failed, but your idea is > better). > I'm guessing this doesn't add too much load to Kamailio. > For learning/understanding the dispatcher... was what I wanted to do possible? Please, keep the mail within the maillist thread. No, it doesn't add too much load to Kamailio. This is one of the major purposes of a transaction stateful proxy (the serial forking in order to implement failover). Take a look to the LCR module. It gives you load balancing (with priorities if you want) and failover features. It's all you need in your scenario. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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call not being accounted when rejected by global blacklist
by Panagiotis Skoulikaritis 15 Apr '10

15 Apr '10

Dear list members I'm using kamailio 3 and the problem that I'm facing is that when a call is rejected by global blacklist, kamailio does not generate an accounting record. Is this normal behavior or I need to configure something somewhere ? Below are parts of the config file # ----- acc params ----- modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "failed_transaction_flag", 3) #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://username:password@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif ---------------------- if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } ------------------------- if (is_method("INVITE")) { setflag(1); # do accounting setflag(2); # log missed calls setflag(3); # log failed calls setflag(4); # do siptrace setflag(5); # log dialog ------------- route[name] { # --- Remove # or %23 from rU --- bm_start_timer("hash_remove"); if($rU =~ "^.*%+") { xlog("------- The number contains %23 ---- "); $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); xlog(" ------- The perl $rU ------- "); } if($rU =~ "^.*#+") { xlog("------- The number contains # ----"); $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); xlog("------- The perl $rU ------- "); } bm_log_timer("hash_remove"); # --- PDT section ------ get service prefix to be appended before is send out bm_start_timer("pdt_lookup"); if(!prefix2domain("2", "0")) { sl_send_reply("404", "Prefix not found"); exit; } $var(dstgrp) = 0; $var(dstgrp) = $(rd{s.int}); if($var(dstgrp)==0) { sl_send_reply("404", "Service not found"); exit; } $avp(s:service_prefix) = $var(dstgrp); xlog("--$rU----------- $avp(s:service_prefix) --- PDT ---------\n"); bm_log_timer("pdt_lookup"); # --- DialPlan section --- get the routing group id xlog("----- The rU is $rU before DP -----\n"); bm_start_timer("dialplan_lookup"); if(dp_translate("1", "$rU/$rU")) { $avp(s:dstgrp) = $(var(dpattrs){s.int}); xlog("----- The destination Drouting GroupID is $avp(s:dstgrp) -----\n "); } bm_log_timer("dialplan_lookup"); xlog("----- The rU is $rU after DP -----\n"); # --- Global Blacklist section --- bm_start_timer("blacklist_lookup"); if(!check_blacklist("globalblacklist")){ xlog("----- The number is blacklisted globaly! -----"); sl_send_reply("403", "Forbidden"); exit; } bm_log_timer("blacklist_lookup"); # --- Dynamic routing section --- route the call based on the routing group id bm_start_timer("drouting_lookup"); if (!do_routing("$avp(s:dstgrp)")) { xlog("----- DRouting don't match dstgrp $avp(s:dstgrp) -----\n"); sl_send_reply("404","Route Not Found -$rU-"); exit; } bm_log_timer("drouting_lookup"); xlog("-----DRouting with $rU and $avp(s:dstgrp) -and $avp(s:service_prefix)----\n"); # Append service prefix in front of number $ru = "sip:" + $avp(s:service_prefix) + $rU + "@" + $rd; xlog(" ------ $ru ----$rU- $avp(s:user)---\n"); t_on_failure("FAIL_ONE"); route(RELAY); exit; return; } Regards Panagiotis

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sr-users April 2010