sr-users March 2016

sr-users@lists.kamailio.org

88 participants
127 discussions

Kamailio 5.0 - Better way of dealing with stateless CANCELs

by Alex Balashov

Currently (AFAIK), restarting Kamailio amidst production call processing is basically "safe"; except for no availability during the few seconds it takes to restart (which will just result in retransmissions until it is available), most things will happen "correctly" after restart even though TM state has been lost: (1) Initial requests will be routed as initial requests always were; (2) In-dialog requests will be loose-routed as sequential requests always were; (3) Replies to open transactions will fall back to stateless routing but will be delivered correctly to their destinations based on SIP fundamentals (i.e. Via). (4) rtpproxy & rtpengine control messages are grouped by Call-ID, so also stateless. If the proper destroy/remove functions are not called from failure_route[] due to lack of TM state, it's not so bad; rtpproxy & rtpengine will see an RTP timeout after a while and expire the bindings on their own. If dialog state is used, it will be lost, but assuming one is willing to live with that, it's okay. I don't know if there has been any work done to create a persistence layer for dialog that can be re-read completely on startup, and if it actually works - does it? - but it's a relatively small price to pay if it's important to integrate a change into production in the middle of the day. The one exception is CANCEL handling. CANCEL is a special animal, since it's a hop-by-hop (branch-level) request, so CANCELs sent from a caller apply to the 'caller -> Kamailio' branch. Kamailio generates separate CANCELs endogenously for one or more 'Kamailio -> gateway' branches. Stateful CANCEL handling with TM is implemented using t_check_trans() or t_relay_cancel(). For example, in the stock config[1] # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } Or, as in our case, more folklorically: if(is_method("CANCEL")) { if(!t_relay_cancel()) { # Corresponding INVITE transaction found, but error # occurred. sl_send_reply("500", "Internal Server Error"); exit; } # Corresponding INVITE transaction for CANCEL was not # found. exit; } In both cases, the corresponding INVITE transaction must exist. Unfortunately, there's no good alternative. According to RFC 3261 Section 16.11 ("Stateless Proxy"): Stateless proxies MUST NOT perform special processing for CANCEL requests. They are processed by the above rules as any other requests. So, in other words, route_logic(CANCEL) == route_logic(initial INVITE). Sometimes, this is possible - with considerable config logic labour - but other times the path taken by the CANCEL is not so deterministic, as for example with round-robin load balancing, random distribution, complex LCR, etc. One is basically left in these situations with the choice of implementing one's own Call-ID/branch => destination persistent state database of some kind, which is, to put it mildly, complicated and undesirable. Now, if the INVITE transaction receives a final negative reply, this will get back to the calling UAC, and it will process it correctly. However, some calls get answered with 2xx. Many UACs will behave reasonably in this situation: when they don't receive a 200 OK for their CANCELs but later receive an answer, they will go ahead and send the end-to-end ACK, then BYE the call. However, they cannot be reliably counted upon to do this. Some simply get drop the INVITE transaction after their CANCEL has gone unreplied for a short time, regardless of whether they receive a final negative reply for it. Is there a better way? Perhaps a feature can be devised by which Kamailio keeps some kind of lightweight and restart-persistent map to which to send the CANCELs? Or perhaps TM is due for a feature that allows the shm transaction table to be dumped to disk and persisted across restarts? Comments welcome. Also, if I'm missing something, please let me know! -- Alex [1] https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L466 -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

8 years, 11 months

Missing pipelimit values in data returned by JSONRPC-S

by Brooks Bridges

When trying to pull a list of pipes and their current values by using pl.dump via the JSONRPC-S module, I am only getting the last result in the reply as opposed to an array. If I use the CLI RPC interface I get the data I expect. Here's a sanitized example: [root@server1 ~]# kamcmd pl.stats PIPE: id=user_1 load=0 counter=29 PIPE: id=user_2 load=0 counter=108 PIPE: id=user_3 load=0 counter=18 PIPE: id=user_4 load=0 counter=0 PIPE: id=user_5 load=0 counter=1 PIPE: id=user_6 load=0 counter=2 PIPE: id=user_7 load=0 counter=0 PIPE: id=user_8 load=0 counter=84 PIPE: id=user_9 load=0 counter=26 PIPE: id=user_10 load=0 counter=0 PIPE: id=user_11 load=0 counter=23 PIPE: id=user_12 load=0 counter=9 PIPE: id=user_13 load=0 counter=23 PIPE: id=user_14 load=0 counter=0 PIPE: id=user_15 load=0 counter=107 PIPE: id=user_16 load=0 counter=10 PIPE: id=user_17 load=0 counter=28 PIPE: id=user_18 load=0 counter=45 PIPE: id=user_19 load=0 counter=7 PIPE: id=user_20 load=0 counter=0 PIPE: id=user_21 load=0 counter=12 PIPE: id=user_22 load=0 counter=94 PIPE: id=user_23 load=0 counter=14 PIPE: id=user_24 load=0 counter=7 PIPE: id=user_25 load=0 counter=18 PIPE: id=user_26 load=0 counter=46 PIPE: id=user_27 load=0 counter=45 PIPE: id=user_28 load=0 counter=50 PIPE: id=user_29 load=0 counter=2 PIPE: id=user_30 load=0 counter=2 [root@server1 ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "pl.stats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","result":"PIPE: id=user_30 load=0 counter=2"} [root@server1 ~]# The jsonrpc config and event route is about as basic as you can get: modparam("jsonrpc-s", "pretty_format", 0) modparam("jsonrpc-s", "transport", 1) event_route[xhttp:request] { xlog("L_INFO", "[R-HTTP-REQUEST:$si] -> Got HTTP request\n"); if($hu =~ "^/jsonrpc") { xlog("L_INFO", "[R-XHTTP-REQUEST:$si] -> Handling JSON RPC request: $hu\n"); jsonrpc_dispatch(); } else { xhttp_reply("404", "Not Found", "text/html", ""); } } Should the JSONRPC-S module be returning this as an array instead of just a single value? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

8 years, 11 months

Removing Via and Record-Route headers

by Marrold

Hi, Is there any way to remove Via and Record-Route headers from requests sent to an endpoint and update the contact header, but have Kamailio statefully remember where the replies need to route? I imagine this would involve mangling the packets slightly to behave similar to a B2BUA. My motivations, in order of priority - 1) Reduce UDP packet size to avoid MTU limitations 2) Improve interoperability, don't expect end points to be able to properly parse / process many via / record-route headers 3) Hide topology. If someone could point me in the right direction in terms of configuration or modules, that would be great. Thanks

8 years, 11 months

What to read after RFC 3261 to get started with Kamailio.

by Oivvio Polite

I want to get started with Kamailio. So far I've understood that I really need to grok SIP before tackling Kamailio so I'm reading the book "SIP demystified" as the starting point of my Kamailio journey. After that I will read RFC 3261. After that I think it's time to read something that will give me good understanding of how to actually implement applications with Kamailio. Searching Amazon for Kamailio returns "Building Telephony Systems with OpenSER" as the results that seems to be most specific to Kamailio. This book is from 2008. Is it still relevant or is it better to go with the online docs? regards, Oivvio

8 years, 11 months

problem with relaying after t_susped() and t_continue()

by Uri Shacked

I have a problem when using t_suspend() and t_continue(). Let's say I would like to have a 5 sec delay before relaying an INVITE. For example the following code: * route[MYROUTE] {* * t_newtran();* * if (is_method("INVITE")) {* * t_reply("100", "Trying");* * t_reply("180", "In Queue");* * $sht(a=>$ci::t_index) = $T(id_index);* * $sht(a=>$ci::t_label) = $T(id_label);* * $sht(a=>$ci::t_suspended) = 1;* * xlog("L_WARN","SUSPENDING t_index = $sht(a=>$ci::t_index) and t_label = $sht(a=>$ci::t_label)");* * t_suspend();* * sleep(5);* * if (t_is_canceled()) drop;* * $sht(a=>$ci::t_suspended) = 0;* * t_continue("$avp(t_index)","$avp(t_label)","1");* * }* * route(RELAY);* * exit;* * }* * route[1] {* * xlog("L_WARN","After transaction continue\n");* * route(RELAY);* * exit;* * }* When using this code, I get the following error: Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: WARNING: <script>: After transaction continue Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: <core> [resolve.c:1693]: sip_hostport2su(): could not resolve hostname: "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [ut.h:319]: uri2dst2(): failed to resolve "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: BUG: tm [t_lookup.c:1481]: t_unref(): tm: t_unref: REQ_ERR DELAYED should have been caught much earlier for 0x7fd4e727a468: 27 (hex 1b) If I try to t_continue() without any route: t_continue("$avp(t_index)","$avp(t_label)",""); The call is working great and the following error pops up: Mar 9 17:10:36 mysip /usr/local/sbin/kamailio[5830]: ERROR: tmx [tmx_mod.c:635]: w_t_continue(): empty action lists in route block [] Am I doing something wrong? My other question is regarding CANCEL. Let's assume I suspended the INVITE and in these 5 sec I received CANCEL for it. Is there a short and elegant way to remove the transaction from suspension? For now I do: * if (is_method("CANCEL")) {* * if (t_check_trans()) {* * if($sht(a=>$ci::t_suspended) == 1) {* * t_continue("$sht(a=>$ci::t_index)","$sht(a=>$ci::t_label)","");* * xlog("L_WARN","After transaction continue\n");* * }* * route(RELAY);* * }* * exit;* * }* Thanks, Uri

8 years, 11 months

LCR prefix based

by Roberto Innaimi

Hi to all, i want to use prefix based LCR routing, but for some reason, something is not working as expected. Can anyone help me? This is the LCR configuration: #LCR MODULE loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://kamailioro:kamailioro@localhost/kamailio") modparam("lcr", "lcr_gw_table","lcr_gw") modparam("lcr", "id_column", "id") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "hostname_column", "hostname") modparam("lcr", "port_column", "port") modparam("lcr", "params_column", "params") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "strip_column", "strip") modparam("lcr", "tag_column", "tag") modparam("lcr", "flags_column", "flags") modparam("lcr", "defunct_column", "defunct") modparam("lcr", "lcr_rule_table", "lcr_rule") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "request_uri_column", "request_uri") modparam("lcr", "stopper_column", "stopper") modparam("lcr", "enabled_column", "enabled") modparam("lcr", "lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR route\n"); } if(method=="INVITE") { xlog("We got an invite"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'\n"); xlog("Domain of destination: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gateway\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); } } } AND this is the error which i have got: 17(7483) ERROR: <script>: We got an invite17(7483) ERROR: <script>: GW Selected '1|sip:|3|||2873019861||5060||;transport=udp|1' 17(7483) ERROR: <script>: Domain of destination: <null> 17(7483) ERROR: <script>: To URI: sip:001326212223313@10.10.0.10 17(7483) ERROR: <script>: Calling the first matched gateway 17(7483) ERROR: <script>: ruri_user_avp: '001326212223313' 17(7483) ERROR: <script>: To URI after next_gw: sip: 001326212223313(a)10.10.0.10 17(7483) ERROR: <script>: Request URI: 326212223313 -- Roberto Innaimi roberto.innaimi(a)gmail.com

8 years, 11 months

Problem with Freeswitch with IP public and Kamailio behind NAT

by Hai Bui Duc Ha

Hi all, I work on Kazoo system. I have NAT problem when configure Freeswitch and Kamailio. As the old network diagram: <https://lh3.googleusercontent.com/-dVJit4UPAL8/VuI4z1LBdKI/AAAAAAAABDo/zSG0…> I configure dispatcher to FS 1,2 with IP private and map IP public <-> IP private for each server via Router. This work fine. Then now, I want to improve FS performance so that I set up new FS with IP public: <https://lh3.googleusercontent.com/-YZl8d3HBo8E/VuI7ChgTGzI/AAAAAAAABD0/AQQW…> I configure dispatcher FS3 with IP public, connection between FS3 & ecallmgr via public IP is Okay but have problem with NAT: *When Kamailio foward SIP INVITE message to FS3 then FS3 response 200 OK message but it send to Ip private of Kamailio not IP public.* As Kamailio I have already configure advertise IP public: local.cfg listen=UDP_SIP advertise 125.212.212.40:5060 listen=TCP_SIP advertise 125.212.212.40:5060 View console log on Freeswitch, I see the ecallmgr response to FS3: 1. bridge({outbound_redirect_fatal="false",call_timeout=20, originate_timeout=20,local_var_clobber="true"}[ecallmgr_Authorizing-ID= "71310939684a2c8acc4e0e6dff1be6c2",ecallmgr_Owner-ID= "188cd101dcea060d080261b328fdf1ca",ecallmgr_Account-ID= "3f221d1ce6959ec04acd372923e253b9",sdp_secure_savp_only="false", sip_invite_domain="hahai1412.anttel-pro.ab-kz-02.antbuddy.com", presence_id="sharklasers(a)hahai1412.anttel-pro.ab-kz-02.antbuddy.com <https://happy.antbuddy.com/sharklasers@hahai1412.anttel-pro.ab-kz-02.antbud…> ",sip_h_X-KAZOO-AOR=sip:sharklasers@hahai1412.anttel-pro.ab-kz- 02.antbuddy.com,absolute_codec_string="^^:PCMU:VP8",leg_timeout="20", effective_callee_id_number="sharklasers",effective_callee_id_name= "sharklasers ",origination_callee_id_number="sharklasers", origination_callee_id_name="sharklasers ",ecallmgr_Realm=" hahai1412.anttel-pro.ab-kz-02.antbuddy.com",ecallmgr_Username= "sharklasers"]sofia/sipinterface_1/sharklasers@113.161.89.159:5914 <https://happy.antbuddy.com/sharklasers@113.161.89.159:5914>;transport=ws ;*fs_path=sip:10.127.15.36:5060*;lr;received="sip:113.161.89.159:5914 ;transport=ws") I research fs_path parameter: https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path I guess this is broken It should be Public IP: *fs_path=sip:125.212.212.40:5060*. But I don't know how I can change this parameter ! Anybody faced that problem ? Any solution ? Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

8 years, 11 months

Kamailio behind NAT

by Safdar Khan

Hello I have been running kamailio on local network including rtpengine and freeswitch. Now i am configuring kamailio with my public IP(trying with stock kamailio configuration to get started) #define WITH_NAT listen=udp:192.168.3.32:5060 listen=tcp:192.168.3.32:5060 advertised_address= "122.xx.xx.xx" advertised_port="5060" and also using rtpengine module at udp:192.168.3.32:22222 now the problems i am facing are 1.> UA can't register with UDP transport(not even recieving any packets at server ,testing with IMSdroid). 2.>TCP is registering the user, but users can't make calls . i've also noticed this error in /var/log/syslog when tcp ca't make calls ERROR: <core> [tcp_main.c:4250]: tcpconn_main_timeout(): connect 100.68.181.111:50842 failed (timeout) please help me out

8 years, 11 months

srv_query question

by Vasiliy Ganchev

Hi folk! I am trying to implement srv_query usage in my script (server is based on Kamailio+Asterisk). The task: when outgoing REGISTER comes from Asterisk (outgoing trunk to operator), and R-URI consists domain name, need to store all ip-addresses of remote side (for following use) Currently, this works using dns_query, and storing in htable structure with remote IPs. But if remote side use SRV, I do not have all IPs Trying to use srv_query function from ipops, I am stuck with following: - srv_query(srvcname, pvid) to get srvcname - for example, "_sip._tcp.example.com", I need to provide, NAPTR record lookup (that is according to RFC3263, and Kamailio does this in core, while request processing) but I see no way how to make NAPTR lookup from script (not involving non Kamailio tools) Looking in mail list archive, found nothing that could help. What I am missing, is there an easy solution, how to cope with my case? Would appreciate any suggestion or idea. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/srv-query-question-tp146546.html Sent from the Users mailing list archive at Nabble.com.

8 years, 11 months

PostgreSQL - How To

by Barış Şekerciler

Hello everyone,I have Kamailio server which runs at 192.168.2.170 address and have PostgreSQL server which runs at 192.168.2.174.So I tried to arrange Kamailio for working with PostgreSQL but I think I give something miss. I couldnt handle it.Do you have any tutorial for this? Or can you tell me how can I exactly do this?Regards.

8 years, 11 months

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sr-users March 2016