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Documentation -> Tutorials -> Load Balancing 1.9This page has been visited 22239 times. Table of Content (hide) This tutorial applied for OpenSIP versions 1.9 1. Load Balancing in OpenSIPSThe "load-balancing" module comes to provide traffic routing based on load. Shortly, when OpenSIPS routes calls to a set of destinations, it is able to keep the load status (as number of ongoing calls) of each destination and to choose to route to the less loaded destination (at that moment). OpenSIPS is aware of the capacity of each destination - it is pre-configured with the maximum load accepted by the destinations. To be more precise, when routing, OpenSIPS will consider the less loaded destination not the destination with the smallest number of ongoing calls, but the destination with the largest available slot. Also, the "load-balancing" (LB) module is able to receive feedback from the destinations (if they are capable of). This mechanism is used for notifying OpenSIPS when the maximum capacity of a destination changed (like a GW with more or less E1 cards). The "load-balancing" functionality comes to enhance the "dispatcher" one. The difference comes in having or not load information about the destinations where you are routing to:
2. Load Balancing - how it worksWhen looking at the LB implementation in OpenSIPS, we have 3 aspects: 2.1 Destination setA destination is defined by its address (a SIP URI) and its description as capacity. Form the LB module perspective, the destinations are not homogeneous - they are not alike; and not only from capacity point of view, but also from what kind of services/resources they offer. For example, you may have a set of Yate/Asterisk boxes for media-related services -some of them are doing transcoding, other voicemail or conference, other simple announcement , other PSTN termination. But you may have mixed boxes - one box may do PSTN and voicemail in the same time. So each destination from the set may offer a different set of services/resources. So, for each destination, the LB module defines the offered resources, and for each resource, it defines the capacity / maximum load as number of concurrent calls the destination can handle for that resource. Example: 4 destinations/boxes in the LB set 1) offers 30 channels for transcoding and 32 for PSTN 2) offers 100 voicemail channels and 10 for transcoding 3) offers 50 voicemail channels and 300 for conference 4) offers 10 voicemail, 10 conference, 10 transcoding and 32 PSTN This translated into the following setup: +----+----------+-------------------------+---------------------------------+ | id | group_id | dst_uri | resources | +----+----------+-------------------------+---------------------------------+ | 1 | 1 | sip:yate1.mycluster.net | transc=30; pstn=32 | | 2 | 1 | sip:yate2.mycluster.net | vm=100; transc=10 | | 3 | 1 | sip:yate3.mycluster.net | vm=50; conf=300 | | 4 | 1 | sip:yate4.mycluster.net | vm=10;conf=10;transc=10;pstn=32 | +----+----------+-------------------------+---------------------------------+ For runtime, the LB module provides MI commands for:
2.2 Invoking Load-balancingUsing the LB functionality is very simple - you just have to pass to the LB module what kind of resources the call requires. The resource detection is done in the OpenSIPS routing script, based on whatever information is appropriated. For example, looking at the RURI (dialed number) you can see if the call must go to PSTN or if it a voicemail or conference number; also, by looking at the codecs advertised in the SDP, you can figure out if transcoding is or not also required. if (!load_balance("1","transc;pstn")) { sl_send_reply("500","Service full"); exit; } The first parameter of the function identifies the LB set to be used (see the group_id column in the above DB snapshot). Second parameter is list of the required resource for the call. A third optional parameter my be passed to instruct the LB engine on how to estimate the load - in absolute value (how many channels are used) or in relative value (how many percentages are used). The load_balance() will automatically create the dialog state for the call (in order to monitor it) and will also allocate the requested resources for it (from the selected box). The resources will be automatically released when the call terminates. The LB module provides an MI function that allows the admin to inspect the current load over the destinations. 2.3 The LB logicThe logic used by the LB module to select the destination is:
Example: 4 destinations/boxes in the LB set 1) offers 30 channels for transcoding and 32 for PSTN 2) offers 100 voicemail channels and 10 for transcoding 3) offers 50 voicemail channels and 300 for conference 4) offers 10 voicemail, 10 conference, 10 transcoding and 32 PSTN when calling load_balance("1","transc;pstn") -> 1) only boxes (1) and (4) will be selected at as they offer both transcoding and pstn 2) evaluating the load : (1) transcoding - 10 channels used; PSTN - 18 used (4) transcoding - 9 channels used; PSTN - 16 used evaluating available load (capacity-load) : (1) transcoding - 20 channels used; PSTN - 14 used (4) transcoding - 1 channels used; PSTN - 16 used 3) for each box, the minimum available load (through all resources) (1) 14 (PSTN) (2) 1 (transcoding) 4) final selected box in (1) as it has the the biggest (=14) available load for the most loaded resource. The selection algorithm tries to avoid the intensive usage of a resource per box. 2.4 Disabling and PingingThe Load Balancer modules provides couple of functionalities to help in dealing with failures of the destinations. The actual detection of a failed destination (based on the SIP traffic) is done in the OpenSIPS routing script by looking at the codes of the replies you receive back from the destinations (see the example at the end of tutorial). Once a destination is detected at failed, in script, you can mark it as disabled via the lb_disable() function - once marked as disabled, the destination will not be used anymore in the LB process (it will not be considered a possible destination when routing calls). For a destination to be set back as enabled, there are two options:
To enable pinging, you need first to set probing_interval to a non zero value - how often the pinging should be done. The pinging will be done by periodically sending a OPTIONS SIP request to the destination - see probing_method option. To control which and when a destination is pinged, there is the probe_mode column in the load_balancer table - see table definition. Possible options are:
2.5 RealTime Control over the Load BalancerThe Load Balancer module provides several MI functions to allow you to do runtime changes and to get realtime information from it. Pushing changes at runtime:
For fetching realtime information :
3. Study Case: routing the media gatewaysHere is the full configuration and script for performing LB between media peers. 3.1 ConfigurationLet's consider the following case: a cluster of media servers providing voicemail service and PSTN (in and out) service. So the boxes will be able to receive calls for Voicemail or for PSTN termination, but they will be able to send back calls only for PSTN inbound. We also want the destinations to be disabled from script (when a failure is detected); The re-enabling of the destinations will be done based on pinging - we do pinging only when the destination is in "failed" status. 4 destinations/boxes in the LB set 1) offers 50 channels for voicemail and 32 for PSTN 2) offers 100 voicemail channels 3) offers 50 voicemail channels 4) offers 10 voicemail and 64 PSTN This translated into the following setup: +----+----------+-------------------------+-------------------+-----------+ | id | group_id | dst_uri | resources | prob_mode | +----+----------+-------------------------+-------------------+-----------+ | 1 | 1 | sip:yate1.mycluster.net | vm=50; pstn=32 | 1 | | 2 | 1 | sip:yate2.mycluster.net | vm=100 | 1 | | 3 | 1 | sip:yate3.mycluster.net | vm=50 | 1 | | 4 | 1 | sip:yate4.mycluster.net | vm=10;pstn=64 | 1 | +----+----------+-------------------------+-------------------+-----------+ 3.2 OpenSIPS Scriptingdebug=1 memlog=1 fork=yes children=2 log_stderror=no log_facility=LOG_LOCAL0 disable_tcp=yes disable_dns_blacklist = yes auto_aliases=no check_via=no dns=off rev_dns=off listen=udp:xxx.xxx.xxx.xxx:5060 # REPLACE here with right values loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/sl/sl.so" loadmodule "modules/db_mysql/db_mysql.so" loadmodule "modules/tm/tm.so" loadmodule "modules/uri/uri.so" loadmodule "modules/rr/rr.so" loadmodule "modules/dialog/dialog.so" loadmodule "modules/mi_fifo/mi_fifo.so" loadmodule "modules/mi_xmlrpc/mi_xmlrpc.so" loadmodule "modules/signaling/signaling.so" loadmodule "modules/textops/textops.so" loadmodule "modules/sipmsgops/sipmsgops.so" loadmodule "modules/load_balancer/load_balancer.so" modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") modparam("dialog", "db_mode", 1) modparam("dialog", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") modparam("rr","enable_double_rr",1) modparam("rr","append_fromtag",1) modparam("load_balancer", "db_url","mysql://opensips:opensipsrw@localhost/opensips") # ping every 30 secs the failed destinations modparam("load_balancer", "probing_interval", 30) modparam("load_balancer", "probing_from", "sip:pinger@LB_IP:LB_PORT") # consider positive ping reply the 404 modparam("load_balancer", "probing_reply_codes", "404") route{ if (!mf_process_maxfwd_header("3")) { send_reply("483","looping"); exit; } if ( has_totag() ) { # sequential request -> obey Route indication loose_route(); t_relay(); exit; } # handle cancel and re-transmissions if ( is_method("CANCEL") ) { if ( t_check_trans() ) t_relay(); exit; } # from now on we have only the initial requests if (!is_method("INVITE")) { send_reply("405","Method Not Allowed"); exit; } # initial request record_route(); # not really necessary to create the dialog from script (as the # LB functions will do this for us automatically), but we do it # if we want to pass some flags to dialog (pinging, bye, etc) create_dialog("B"); # check the direction of call if ( lb_is_destination("$si","$sp","1") ) { # call comes from our cluster, so it is an PSNT inbound call # mark it as load on the corresponding destination lb_count_call("$si","$sp","1", "pstn"); # and route is to our main sip server to send call to end user $du = "sip:PROXY_IP:PORXY_PORT"; # REPLACE here with right values t_relay(); exit; } # detect resources and store in an AVP if ( $rU=~"^VM_" ) { # looks like a VoiceMail call $avp(lb_res) = "vm"; } else if ( $rU=~"^[0-9]+$" ) { # PSTN call $avp(lb_res) = "pstn"; } else { send_reply("404","Destination not found"); exit; } # LB function returns negative if no suitable destination (for requested resources) is found, # or if all destinations are full if ( !load_balance("1","$avp(lb_res)") ) { send_reply("500","Service full"); exit; } xlog("Selected destination is: $du\n"); # arm a failure route for be able to catch a failure event and to do # failover to the next available destination t_on_failure("LB_failed"); # send it out if (!t_relay()) { sl_reply_error(); } } failure_route[LB_failed] { # skip if call was canceled if (t_was_cancelled()) { exit; } # was a destination failure ? (we do not want to do failover # if it was a call setup failure, so we look for 500 and 600 # class replied and for local timeouts) if ( t_check_status("[56][0-9][0-9]") || (t_check_status("408") && t_local_replied("all") ) ) { # this is a case for failover xlog("REPORT: LB destination $du failed with code $T_reply_code\n"); # mark failed destination as disabled lb_disable(); # try to re-route to next available destination if ( !load_balance("1","$avp(lb_res)") ) { send_reply("500","Service full"); exit; } xlog("REPORT: re-routing call to $du \n"); t_relay(); } } 4. Comments |