sr-users February 2012

sr-users@lists.kamailio.org

79 participants
140 discussions

UAC canceling a specific early dialog with BYE
by Andrew Pogrebennyk 21 Feb '12

21 Feb '12

Hello, we have a Kamailio proxy which gets the call from PSTN gw and does some call forwarding (serial forking) to several destinations through our sbc The call flow I am looking at is: - Kamailio sends INVITE to branch_1. - branch_1 sends 180 with to-tag*, proxy relays it to the gw * 180 meets the requirements for dialog creating 18x responses in sections 12.1, 12.1.1 because it contains to-tag, contact and mirrored record-route. - After some seconds Kamailio sends a CANCEL to branch_1. - And sends the INVITE to branch_2. - branch_1 replies 200 for the CANCEL and 487 for the INVITE. - branch_2 replies 180 and 200 for the INVITE. - When PSTN gw receives that it sees it still needs to cancel other early dialog established by 180 from branch_1. - The PSTN gw sends a BYE with to-tag of branch_1 to cancel this specific early dialog. SIP allows early dialogs to individually released while other dialogs continue, as written in RFC, section 15: "The BYE request is used to terminate a specific session or attempted session. In this case, the specific session is the one with the peer UA on the other side of the dialog. (...). The caller’s UA MAY send a BYE for either confirmed or early dialogs, and the callee’s UA MAY send a BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs." The BYE follows the loose routing path, proxy gets 481 from the sbc and forwards that response back to PSTN gw, which somehow breaks it. AFAICS it's not specified in RFC what should the behavior look like when getting both a 200 and error-class response for the INVITE (quotes are most welcome!). IMO it would be more correct to absorb BYE in proxy but I see a big problem here: branch_2 can even ring for 5 minutes and it's not feasible for proxy to have a wt-timer that long. Also it's not possible to inform the gw that early dialog has cleared as soon as we receive 200/487 from branch_1. So I'm not sure which party is at fault and if we can workaround that somehow in the Kamailio. Any thoughts?

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Re: [SR-Users] Kamailio ims / rtpproxy
by trojak＠st.fri.uniza.sk 21 Feb '12

21 Feb '12

Hi All! We did litttle research: - both behind the NAT = all OK = straeming over the rtpproxy - caller behind NAT (other public) = all BAD = no change in SDP (Connection Address) - rtp stream sending on private addres - calle behind NAT (other public) = all BAD = SDP has changed but only private IP address... public IP unchanged = so rtpproxy do not ralay one way stream Can somebody (especialy IMS) please help us Thanks Best regards, Lubo > Hi Daniel, > > thank you for answer. I think that parts of kamailio default config > will not working with pcscf. (e.g. there is on pcscf no database - > it has hss) > > It partialy works with pcscf config -> it rewrites Connecition > Infromation IP in SDP: > - sometimes with IP of rtpproxy (more often when both clietnts are > behind NAT) > - sometimes with public IP of NATed client (we are not sure why) > - or unchanged (when one client is behind NAT and other is public) > > but this works only with Monster client, with Boghe IP is unchanged > (shouldn't it be client independent ?). > > Maybe someone from IMS help us. > > Thanks all for any advice. > > Best regards, > > Lubo > > Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> On 1/23/12 10:43 AM, trojak(a)st.fri.uniza.sk wrote: >>> >>> Hi, >>> >>> I am trying to make kamailio PCSCF working with rtpproxy (for NAT >>> traversal). >>> I would like to ask, if there is any tutorial for this, or if you >>> can give me any advice. >> Maybe IMS guys can give other hints, what is on public so far was >> summarized at: >> * http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ >> >>> I also have installed rtpproxy package; #!define WITH_NAT, and >>> defined rtpproxy port in config file... >>> But this is not working. I noticed, that in default config file of >>> pcscf is nothing in routing logic with rtpproxy (or it is >>> unnecessary ?), so I try to copy route[RELAY] route[NAT] >>> route[RTPPROXY] modules from kamailio (no pcscf) config file... >>> but also with no success. >>> >>> Can you help me please ? What I am doing wrong ? Do you have any >>> configuration working with rtpproxy ? >> Be sure you can the nat handling routes for INVITES behind nat. You >> can follow the logic in the default config file and adapt it for >> you pcscf config. In summary should be like: >> - invite comes in, test if it is from caller behind nat >> - lookup location returns a branch flag if the calee is behind nat >> (you have to set that branch flag for register request coming from >> users) >> - if caller or callee are behind nat, engage rtpproxy for that call >> and set a onreply route to update sdp for 18x/200 replies >> - default config adds a Route header parameter for nated calls to >> detect easily the natted situation for re-invites or bye >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Dialogs not removed from memory, and occasionally persistent in DB as well
by Øyvind Kolbu 20 Feb '12

20 Feb '12

Hi, we use the dialog-module to keep track of concurrent calls, and then set a treshold for simultanious calls. After we upgraded from 1.5 to 3.2, we've been having a lot of dialogs which are stuck in both the database and memory. kamailio -V version: kamailio 3.2.1 (i386/linux) 918035-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 918035 -dirty compiled on 10:00:10 Dec 20 2011 with gcc 4.1.2 Our configuration: modparam("dialog", "dlg_flag", 3) modparam("dialog", "default_timeout", 21600) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", VOIP_DATA4) modparam("dialog", "db_mode", 1) #!ifdef VOIP1 modparam("dialog", "table_name", "dialog1") modparam("dialog", "vars_table_name", "dialog_vars1") #!endif #!ifdef VOIP2 modparam("dialog", "table_name", "dialog2") modparam("dialog", "vars_table_name", "dialog_vars2") #!endif modparam("dialog", "profiles_with_value", "busy") To Kamailio instances, voip1 and voip2, share the same config, but define VOIP1 and VOIP2, respectively. Then in the main route: if !allow_trusted() { route(AUTH); } if (is_method("INVITE")) { setflag(3); dlg_manage(); if ($avp(s:f_uid) != $null) { set_dlg_profile("busy","$avp(s:f_uid)"); get_profile_size("busy", "$avp(s:f_uid)", "$var(dlg_busy)"); xlog("L_INFO", "BUSY: f_uid: $avp(s:f_uid), dlg_busy: $var(dlg_busy)\n"); } } $avp(s:f_uid) is set in route(AUTH), so it will catch calls originating from one of our users. Then later in the callforward route: # Testing dialog $var(dlg_busy) = 0; $var(dlg_busy1) = 0; $var(dlg_busy2) = 0; #!ifdef VOIP1 get_profile_size("busy", "$rU", "$var(dlg_busy1)"); sql_query("data4","select id from dialog2 where caller_contact like 'sip:$rU@%' or callee_contact like 'sip:$rU@%'", "dialog"); $var(dlg_busy2) = $dbr(dialog=>rows); #!endif #!ifdef VOIP2 get_profile_size("busy", "$rU", "$var(dlg_busy2)"); sql_query("data4","select id from dialog1 where caller_contact like 'sip:$rU@%' or callee_contact like 'sip:$rU@%'", "dialog"); $var(dlg_busy1) = $dbr(dialog=>rows); #!endif sql_result_free("dialog"); $var(dlg_busy) = $var(dlg_busy1) + $var(dlg_busy2); And then check if $var(dlg_busy) is above a users treshold. Finally when routing to a local user we mark it as busy: set_dlg_profile("busy","$rU"); As of writing I have 5 entries in the dialog1-table, all valid, but: # kamctl fifo dlg_list | grep -c "^dialog" 18 # kamctl fifo dlg_list | grep state | sort | uniq -c 13 state:: 1 5 state:: 4 on the other hand in the busy profile, which is the one we use: # kamctl fifo profile_list_dlgs busy | grep -c "^dialog" 11 # kamctl fifo profile_list_dlgs busy | grep state | sort | uniq -c 6 state:: 1 5 state:: 4 It seems as this happens when, for some reason, kamailio is trying to parse duplicate INVITEs. From the logs: # grep "^Jan 17" /var/log/messages | grep BUSY | grep -v "dlg_busy: 1" | cut -d\ -f8- |sort | uniq -c 1 BUSY: f_uid: 2367449, dlg_busy: 2 3 BUSY: f_uid: 2367453, dlg_busy: 2 1 BUSY: f_uid: 2367453, dlg_busy: 3 1 BUSY: f_uid: 2367453, dlg_busy: 4 1 BUSY: f_uid: 2574912, dlg_busy: 2 1 BUSY: f_uid: 2582270, dlg_busy: 2 1 BUSY: f_uid: ahmedma, dlg_busy: 2 1 BUSY: f_uid: josephim, dlg_busy: 2 1 BUSY: f_uid: perla, dlg_busy: 2 1 BUSY: f_uid: rosalinm, dlg_busy: 2 1 BUSY: f_uid: suresha, dlg_busy: 2 1 BUSY: f_uid: sveinb, dlg_busy: 2 I've full debugging logs available, if needed. -- Øyvind Kolbu

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How to deal with diversion loops.
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, the maxfwd will not help me prevent the loop on the first round, it would only stop it a little sooner the X times. no? the dlg_var is interesting, but i didn't understand how to use it exactly. how do i match two dialogs? is the spiral detection on dialog module helps? thanks, Uri >Hi Uri, >for that you can easily count the forwards by using the maxfwd module >(mf_process_maxfwd_deader <your max>) or you can store all your spiraled >calls into a dialog-variable like >$dlg_var(forward-chain)=<B>;<C>|<C>;<B> and check whether the next >number is already stored or not. Bye Sven Am 20.02.2012 13:20, schrieb Uri Shacked: >* *>* Hi, *>* *>* The service i built in kamailio is simple - a caller dials an Access *>* number, my gateway send it to kamailio. *>* *>* kamailio finds the shadow number for this Access number and send an *>* invite to the same gateway with the shadow number as destination *>* (state full proxy). *>* *>* In kamailio i change only the RURI and the “To” and “From” stays *>* untouched. *>* *>* Now, when a client performs “diversion”, I can see that I can find *>* myself in an unwanted long loop…. *>* *>* Example : *>* *>* A calls B *>* *>* B diverted the call unconditionally to C. *>* *>* C is an Access Number that its shadow number is D. *>* *>* D diverted the call to B……. *>* *>* Well, that is a bad loop to have! *>* *>* It can be longer or shorter, but the idea I understood (I think). *>* *>* I thought about diversion check, counters, dialog “To” and “From” *>* check, etc…. *>* *>* Anyone here dealt with it? *>* *>* What will be the best way to check and see if the situation accrues? *>* *>* Thanks, *>* *>* Uri *>* *

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integration kamailio-sems
by Uri Shacked 20 Feb '12

20 Feb '12

Thanks, did you tried to develop an IVR system based on SEMS? will it be better? harder? more flexible than Asterisk?

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Kamailio as SBC
by Martin Schiøtz 20 Feb '12

20 Feb '12

Hi I'm new to be working with SIP. We have a SIP server and it handles SIP well but it does not handle far-end-nat. Can the Kamailio server be used as a SBC (Session Border Controller) to handle far-end-nat? Just forward all SIP packages to a SIP server and fix the far-end-nat problem? Cheers! Martin

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How to deal with diversion loops.
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, The service i built in kamailio is simple - a caller dials an Access number, my gateway send it to kamailio. kamailio finds the shadow number for this Access number and send an invite to the same gateway with the shadow number as destination (state full proxy). In kamailio i change only the RURI and the “To” and “From” stays untouched. Now, when a client performs “diversion”, I can see that I can find myself in an unwanted long loop…. Example : A calls B B diverted the call unconditionally to C. C is an Access Number that its shadow number is D. D diverted the call to B……. Well, that is a bad loop to have! It can be longer or shorter, but the idea I understood (I think). I thought about diversion check, counters, dialog “To” and “From” check, etc…. Anyone here dealt with it? What will be the best way to check and see if the situation accrues? Thanks, Uri

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integration kamailio-sems
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I started reading a little on SEMS. Tried to look for the way Kamailio works has the frontend sip server and SEMS will be the media server (IVR, announcments etc). I didn't understand which kamailio module should i work with in order to open sockets with SEMS. Or do i get it all wrong? Uri

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$time(year) gives 1900
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I am using the $time(name) pv. On my lab machine (centos 5.7) i get the right value (2012). But, on my other machine (centos 6.2) i get 1900. what am i missing here?

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Mtree VS Htable getting huge amount of data from DB not on local machine
by Uri Shacked 20 Feb '12

20 Feb '12

> > Hi, > > untill today i used the mtree module to get some data from a remote DB and > load it into memory. > the table in the data base has 5 million rows. prefix is a number and > value is some string. > > i moved to htable because i need the prefix exact match and not longest.... > > now, after 5-10 sec, i get the following error (that i never got in mtree) > - "lost connection to mysql during query" and kamailio does not start. > > any ideas how to solve it? > > cheers, > > Uri >

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sr-users February 2012