sr-dev October 2024

sr-dev@lists.kamailio.org

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[kamailio/kamailio] Memory usage increases everytime tls.reload is executed (Issue #3823)
by Sebastian Denz 07 Nov '25

07 Nov '25

### Description We are using Kamailio 5.7.4 on Debian 12 (from http://deb.kamailio.org/kamailio57) with rtpengine as an Edgeproxy for our clients. The instance terminates SIP/TLS (with Cliencertificates) and forwards the SIP Traffic to internal systems. After some days we are getting errors like this `tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: shm=7318616 threshold1=8912896` First we thought Kamailio just doesnt have enough memory, so we doubled it.. But after some days the Logmessage (and Userissues) occured again. So we monitored the shmmem statistics and found that used and max_used are constantly growing til it reaches the limit. As i mentioned we are using client-certificates and so we are also using the CRL feature. We do have a systemd-timer which fetches the CRL every hour and runs 'kamcmd tls.reload' when finished. Our tls.cfg looks like this: ``` [server:default] method = TLSv1.2+ private_key = /etc/letsencrypt/live/hostname.de/privkey.pem certificate = /etc/letsencrypt/live/hostname.de/fullchain.pem ca_list = /etc/kamailio/ca_list.pem ca_path = /etc/kamailio/ca_list.pem crl = /etc/kamailio/combined.crl.pem verify_certificate = yes require_certificate = yes [client:default] verify_certificate = yes require_certificate = yes ``` After testing a bit we found that every time tls.reload is executed Kamailio consumes a bit more memory which eventually leads to all the memory being consumed which leads to issues for our users. See following example: ``` [0][root@edgar-dev:~]# while true ; do /usr/sbin/kamcmd tls.reload ; /usr/sbin/kamcmd core.shmmem ; sleep 1 ; done Ok. TLS configuration reloaded. { total: 268435456 free: 223001520 used: 41352552 real_used: 45433936 max_used: 45445968 fragments: 73 } Ok. TLS configuration reloaded. { total: 268435456 free: 222377960 used: 41975592 real_used: 46057496 max_used: 46069232 fragments: 78 } Ok. TLS configuration reloaded. { total: 268435456 free: 221748664 used: 42604992 real_used: 46686792 max_used: 46698080 fragments: 77 } Ok. TLS configuration reloaded. { total: 268435456 free: 221110832 used: 43242408 real_used: 47324624 max_used: 47335608 fragments: 81 } ^C [130][root@edgar-dev:~]# ``` ### Troubleshooting #### Reproduction Everytime tls.reload is called the memory consumptions grows.. #### Debugging Data  ``` If you let me know what would be interesting for tracking this down, i am happy to provide logs/debugging data! ``` #### Log Messages  ``` If you let me know what would be interesting for tracking this down, i am happy to provide logs/debugging data! ``` #### SIP Traffic SIP doesnt seem to be relevant here ### Possible Solutions Calling tls.reload less often or restart kamailio before memory is consumed ;) ### Additional Information ``` version: kamailio 5.7.4 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 12.2.0 ``` * **Operating System**: ``` * Debian GNU/Linux 12 (bookworm) * Linux edgar-dev 6.1.0-20-amd64 #1 SMP PREEMPT_DYNAMIC Debian 6.1.85-1 (2024-04-11) x86_64 GNU/Linux ``` -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3823 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3823(a)github.com>

10 46

[tracker] Assignee added: Assignment operators don't work
by sip-router 29 Oct '25

29 Oct '25

THIS IS AN AUTOMATED MESSAGE, DO NOT REPLY. A user has added themself to the list of users assigned to this task. FS#100 - Assignment operators don't work User who did this - Alex Hermann (axlh) http://sip-router.org/tracker/index.php?do=details&task_id=100 You are receiving this message because you have requested it from the Flyspray bugtracking system. If you did not expect this message or don't want to receive mails in future, you can change your notification settings at the URL shown above.

4 3

[kamailio/kamailio] tls: Create one shared context and reuse it. (PR #3972)
by Xenofon Karamanos 24 Oct '25

24 Oct '25

#### Pre-Submission Checklist    - [x] Commit message has the format required by CONTRIBUTING guide - [x] Commits are split per component (core, individual modules, libs, utils, ...) - [x] Each component has a single commit (if not, squash them into one commit) - [x] No commits to README files for modules (changes must be done to docbook files in `doc/` subfolder, the README file is autogenerated) #### Type Of Change - [ ] Small bug fix (non-breaking change which fixes an issue) - [x] New feature (non-breaking change which adds new functionality) - [ ] Breaking change (fix or feature that would change existing functionality) #### Checklist:  - [ ] PR should be backported to stable branches - [x] Tested changes locally - [x] Related to issue #3823 #### Description  This PR aims to implement what was discussed in [mailing list](https://lists.kamailio.org/mailman3/hyperkitty/list/sr-dev@lists.kama… regarding some `tls.reload` and increasing memory usage. It adds a new parameter `enable_shared_ctx` in `tls` module that if set to 0, preserves the old behavior and if set to 1 (other than 0 tbh), it creates a single SSL context that is being shared. This have the effect of using way less memory when initialized as well, but also minimizes (can't say it fixes the problem) the `tls.reload` memory increase. I have also added a small markdown (comparison.md) file, where some comparisons where made between enabled/disabled shared context and with/without CA file (where the initial problem was occurring by the reporter). Feedback would be necessary to verify whether this patch, acts as expected and kamailio works as intented. You can view, comment on, or merge this pull request online at: https://github.com/kamailio/kamailio/pull/3972 -- Commit Summary -- * tls: Add parameter for shared contexts * tls: Comparison for enable_shared_ctx -- File Changes -- A comparison.md (15) M src/modules/tls/tls_domain.c (172) M src/modules/tls/tls_mod.c (11) M src/modules/tls/tls_mod.h (1) -- Patch Links -- https://github.com/kamailio/kamailio/pull/3972.patch https://github.com/kamailio/kamailio/pull/3972.diff -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/pull/3972 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/pull/3972(a)github.com>

4 17

[kamailio/kamailio] core parsing of SIP message failed (Issue #3062)
by soroshsabz 11 Sep '25

11 Sep '25

ITNOA ### Description I try to connect Kamailio and Freeswitch together, and initiate call from Zoiper5 soft phone through Kamailio My Kamailio config is below ``` #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:172.28.72.99:5060 advertise 172.28.72.99:5060 #listen=tcp:172.28.72.99:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=no /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); # handle requests within SIP dialogs # route(WITHINDLG); ### only initial requests (no To tag) # authentication # route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); #if (is_method("INVITE|SUBSCRIBE")) { # record_route(); #} # account only INVITEs #if (is_method("INVITE")) { # setflag(FLT_ACC); # do accounting #} # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests #route(PRESENCE); # handle registrations # route(REGISTRAR); #if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} # dispatch destinations to PSTN #route(PSTN); # user location service #route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE #!ifdef WITH_MSGREBUILD # apply changes in case the request headers or body were modified msg_apply_changes(); #!endif if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-Hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # JSONRPC over HTTP(S) routing #!ifdef WITH_JSONRPC event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ``` As you can see I just one thing in `request_route` ``` request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); ``` The `172.28.69.52` is ip of my Freeswitch After I try to call from Zoiper5, I get below error in Kamailio log and Zoiper does not work. ``` 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) ``` ### Troubleshooting #### Reproduction * pull official Kamailio docker image `kamailio/kamailio 5.4.6-buster` * then run below command ``` docker run --net=host --name kamailio \ -v /home/ssoroosh/kamailio:/etc/kamailio \ kamailio/kamailio:5.4.6-buster ``` #### Debugging Data I do not have any debug data #### Log Messages ``` Listening on udp: 172.28.72.99 [172.28.72.99]:5060 advertise 172.28.72.99:5060 Aliases: 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2919]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 13(19) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/19 15(21) INFO: ctl [io_listener.c:210]: io_listen_loop(): io_listen_loop: using epoll_lt as the io watch method (auto detected) 2(8) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<�> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<�> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) ``` #### SIP Traffic My pcap is attached [sip.zip](https://github.com/kamailio/kamailio/files/8326814/sip.zip) ### Possible Solutions I have not any solution yet. ### Additional Information * **Kamailio Version** - output of `kamailio -v` ``` version: kamailio 5.4.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 ``` * **Operating System**: ``` Linux master 5.4.0-104-generic #118-Ubuntu SMP Wed Mar 2 19:02:41 UTC 2022 x86_64 GNU/Linux root@master:/home# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster ``` -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3062 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3062(a)github.com>

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[kamailio/kamailio] [Websocket] websocket:closed sometimes is not fired (Issue #3950)
by Gustavo Almeida 27 Jun '25

27 Jun '25

### Description After carrying out some load tests using the `Websocket` module, I noticed that sometimes the `websocket:closed` event is not triggered. This problem seems to be particularly noticeable when the load is high and simultaneous. Example: 1000 records simultaneously. In the tests carried out, I realized that the number of times the event is not triggered is small, in the order of [1-10]/1000. I check that when I do a `ws.dump` I don't have any active connections, and I confirm that they are all effectively cleaned (`netstat -ano | grep 8061`). However, the event is not triggered for some connections. I can see this in the logs, and also when i display the content of an hash table, i'm dropping key inside `websocket:closed` event. ### Troubleshooting The troubleshooting was simplistic, i.e. I ran the same test n times, and validated that the event was triggered for all connections, the number of `ws` connections listed by `Kamailio`, and the open connections. #### Reproduction To reproduce, we can use any mechanism that emulates the use of the `Websocket` module. For example, I used the `sipexer` tool to create a `wss` connection with `Kamailio`, forcing simultaneous registrations. Example command: `sipexer -cb -mt register -ex 60 -au <user> -ap <pass> -fuser example -fd mydomain -wso https://10.0.0.12:10000 -ruri sip:kamailio -su wss://kamailio:8061` Kamailio TLS listener: `listen=tls:10.0.0.12:8061 advertise PUBLIC_IP` Websocket module: ``` loadmodule "websocket.so" tcp_accept_no_cl=yes ``` ``` event_route[websocket:closed] { xlog("L_NOTICE", "WebSocket connection closed $proto:$si:$sp\n"); <I also delete where a htable key> } ``` #### Debugging Data ``` [root@ ~]$ prcmd ws.dump { connections: { } info: { wscounter: 0 truncated: no } } [root@ ~]$ netstat -ano | grep 8061 tcp 0 0 10.0.0.12:8061 0.0.0.0:* LISTEN off (0.00/0/0) [root ~]$ prcmd htable.dump wsauth | grep "name" | wc -l 3 [root ~]$ cat /var/log/kamailio.log | grep "WebSocket connection closed" | wc -l 997 [root ~]$ cat /var/log/proxy-registrar/proxy-registrar.log | grep "ERR" | grep "WAR" | wc -l 0 ``` #### Log Messages Nothing relevant. #### SIP Traffic Nothing relevant here, the SIP Flow is OK. ### Additional Information ``` version: kamailio 5.6.4 (x86_64/linux) a004cf-dirty flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a004cf -dirty compiled on 23:17:27 Jun 7 2024 with gcc 4.8.5 ``` * **Operating System**: ``` CentOS Linux release 7.9.2009 (Core) ``` -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3950 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3950(a)github.com>

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[kamailio/kamailio] Writing to $fU or $tU appends to header user (Issue #3165)
by Benoît Panizzon 24 Jun '25

24 Jun '25

Dear Gang Possibly @oej could provide more in-depth information as he has witnessed this issue. Usually the user of the from URI is the phone number displayed at the destination. There are situations where this phone number is translated. As example. In Switzerland, the user is used to see numbers in a local format. National number starting with 0 and international numbers with 00 but on interconnection between telcos, e164 is used. So basically when a call is sent to a customer '+41' is replaced by '0' and '+' is replaced by '00'. Let's start with an example From: header: `From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone` So shortly before the call is sent out to the location of the registered CPE, this is done: ``` if ($fU =~ "^\+41") { $fU = "0" + $(fU{s.substr,3,0}); } else if ($fU = ~ "^\+") { $fU = "00" + $(fU{s.substr,1,0}); } ``` What is sent to the CPE now looks like this: `From: "Maurice Moss" <sip:0991234567@example.com>;user=phone` Now we hit an error like 486 BUSY and the destination has call forwarding active to a mobile phone on another TSP. So we have to send the call out back the IC and numbers need to be translated back to e164. We handle this in a failure route, which in turn could trigger a branch route. So we revert the number back to e164: `$fU = "+41" + $(fU{s.substr,1,0});` Expected outcome: `From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone` Observed outcome: `From: "Maurice Moss" <sip:0991234567+41991234567@example.com>;user=phone` So setting $fU more than once is appending to the user element of the From header URI. This behavior has not been found in any documentation. I have been working around most of the issues by making sure I change $fU (and $tU) at the latest possible time and only once. But in the case described above, I have not been able to come up with a work-around yet. I also can't think of any benefit of the way those PV are handled or any harm that could be done, to handle them differently and make the last 'write' overwrite and previous value, instead of appending. Thank you for looking into this. -Benoît- -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3165 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3165(a)github.com>

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[kamailio/kamailio] tcp_timer_check_connections() logs on CRITICAL and is not caught by event_route[tcp:timeout] (Issue #3995)
by Fritjof Höst 13 Jun '25

13 Jun '25

### Description Hi! We are in the progress of upgrading from Kamailio 5.5 to 5.8. During our testing we have noticed a new error being reported from Kamailio. We don’t see any other errors following it. ```jsx /usr/sbin/kamailio[201]: CRITICAL: <core> [core/tcp_main.c:5544]: tcp_timer_check_connections(): message processing timeout on connection id: 67896 (state: 3) - closing ``` It does seem to be [new code](https://github.com/kamailio/kamailio/blob/master/src/core/tcp_main.c#… in Kamailio reporting this issue. Given that this is a fairly expected thing, cleaning up a connection which receives no traffic within the given time, is there a need for it to be reported on CRITICAL? I’d also expect it to be caught by ``` event_route[tcp:timeout] { xlog("L_INFO","connection $conid timeouts (unanswered keepalives)"); } ``` given that [the description](https://www.kamailio.org/docs/modules/stable/modules/tcpops.ht… of this one is `Called for connection timeouts (unanswered keepalives).`. ### Troubleshooting We don't have any way to reproduce it, we are still investigating it to figure out the cause. It happens around every 2 hours, so we think there might some some scheduled test or job running in our test system causing this. ### Additional Information * **Kamailio Version** - 5.8.3 * **Operating System**: Debian 12 AMI from the AWS Marketplace. -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3995 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3995(a)github.com>

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[kamailio/kamailio] Possibly adjusting the SDP parser to be more lenient towards scenarios where a stream is being removed/rejected and connection-information "c=" is only available per media (no session-wide "c=") and the removed/rejected stream omits all fields (Issue #3982)
by thomas414 31 May '25

31 May '25

### Description I encountered an issue with a specific device that is receiving a T.38 fax. The call gets established towards the device normally with inband media, then the device requests T.38 media and tries to remove the audio media via sending port 0 in the ReInvite SDP. Also it sends connection-information only on the media level and there is no "c=" on the session level. Reading through the related RFCs and the kamailio code and discussing this on the mailing list, we came to the conclusion that kamailio is doing everything correct as per RFC. Though we were wondering if it would be possible to adjust the parser to be not as strict as it is right now, and allow a missing "c=" line on the media-level if the stream is removed/rejected via port 0, since i see no sense in requiring connection information. Of course we are also in contact with the vendor to hopefully adjust on their side. Here are the related RFCs: 1. RFC8866 5.7 `A session description MUST contain either at least one "c=" line in each media description or a single "c=" line at the session level. It MAY contain a single session-level "c=" line and additional media-level "c=" line(s) per-media-description, in which case the media-level values override the session-level settings for the respective media.` 2. RFC3264 8.2 `Existing media streams are removed by creating a new SDP with the port number for that stream set to zero. The stream description MAY omit all attributes present previously, and MAY list just a single media format.` At first i was especially confused by the RFC3264 8.2 part, since it seemed correct what the device is sending, but if you read carefully and keep the wording for SDP in mind only attributes ("a=") MAY be omitted. So a "c=" line should still be in the SDP for the removed media if it's not included on the session-level. Or do you see this differently? ### Expected behavior Kamailio allows and parses the SDP when there is no session-wide "c=", a media stream is being removed via port zero and there is no "c=" for this media stream and only the remaining media streams include a "c=" line. #### Actual observed behavior Kamailio throws an error when trying to parse the SDP. #### Log Messages ``` <core> [core/parser/sdp/sdp.c:523]: parse_sdp_session(): can't find media IP in the message ``` #### SIP Traffic ``` v=0 o=xmserver 1726638425 1726638427 IN IP4 169.254.1.1 s=xmserver t=0 0 m=audio 0 RTP/AVP 8 m=image 56002 udptl t38 c=IN IP4 169.254.1.1 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy ``` ### Possible Solutions ### Additional Information * **Kamailio Version** - output of `kamailio -v` ``` version: kamailio 5.8.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 12.2.0 ``` * **Operating System**: ``` Debian 12 Linux hostname 6.1.0-20-amd64 #1 SMP PREEMPT_DYNAMIC Debian 6.1.85-1 (2024-04-11) x86_64 GNU/Linux ``` -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3982 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3982(a)github.com>

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[kamailio/kamailio] Inactive video stream: can't find media IP in the message (Issue #3406)
by schoberw 31 May '25

31 May '25

### Description In some situations the other side UA must reply with an inactive media stream. E.g. if video was added but not supported by the other side. The UAS then must answer with an m=video 0 RTP/.. line. It seams that the video section must contain a c= line with an IP address for rtpengine to function. If there is no c line (no IP address) the SDP body cannot be parsed and thus no RTP proxy is invoked. ### Troubleshooting Not easy to reproduce since the answer must bei "wrong". Is it correct to reply an SDP body (media = video) like this? ``` m=video 0 RTP/AVPF 96 a=label:1 a=inactive a=mid:1 ``` Maybe not (but Audiocodes SBC lates LTS version does it) - so we are dependent on other side now that this service works. #### Reproduction reInvite with Video to a UA that has no video Support (e.g. Bria -> Snom). Drop the c line in the video part with some textops in 200 OK (just to reproduce of course). A -> B, connect the call A -> + video, B has no video support A presses hold #### Log Messages Incoming 200 OK after doing hold with an inactive video: ``` v=0 o=root 436040161 309284577 IN IP4 1.2.3.5 s=call t=0 0 m=audio 14200 RTP/AVPF 9 8 126 c=IN IP4 1.2.3.5 a=ptime:20 a=recvonly a=mid:0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-15 m=video 0 RTP/AVPF 96 a=label:1 a=inactive a=mid:1 Mar 27 17:52:22 c5p05es2-1 /usr/sbin/kamailio[2938205]: ERROR: <core> [core/parser/sdp/sdp.c:490]: parse_sdp_session(): can't find media IP in the message Mar 27 17:52:22 c5p05es2-1 /usr/sbin/kamailio[2938205]: INFO: <script>: >>> Sending Reply: 200 OK (1.2.3.4:443 -> 100.108.48.38:65251) ``` Compare to working SDP parsing: ``` a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-15 m=video 0 RTP/AVP 96 c=IN IP4 1.2.3.5 a=inactive a=mid:1 Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: ONREPLY_ROUTE[FORWARD] Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: ROUTE[RTPP_REPLY] Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > 200 with video Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > 200 with inactive video Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > Remove inactive video Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > Answer to WebRTC client: 200 - qijpuibkv6ufhnd282ks Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: >>> Sending Reply: 200 OK (91.237.65.14:443 -> 80.108.48.38:65388) ``` ### Possible Solutions Tried to fix in script: ``` if(sdp_with_media("video")) xlog("L_INFO", "> $rs with video"); if(sdp_with_active_media("video")) xlog("L_INFO", "> $rs with active video"); if(!sdp_with_active_media("video")) xlog("L_INFO", "> $rs with inactive video"); if(sdp_with_media("video") && !sdp_with_active_media("video")) { # try fix for RMT#60189 xlog("L_INFO", "> Remove inactive video"); sdp_remove_media("video"); msg_apply_changes(); # needed? } ``` But since sdpops cannot parse it cannot remove the buggy media, too. See the log: non of the xlog is logging. If there is inactive media the c line is not mandatory. The parser should accept this by adding a default IP of 0.0.0.0, if needed Doing tricks with textops is not a easy nor performant solution to accept this wrong SDP from other parties. ### Additional Information ``` version: kamailio 5.6.4 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 ``` * **Operating System**: Debian Bullseye -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3406 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3406(a)github.com>

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[kamailio/kamailio] core: local TCP socket is bound on listening address (PR #3925)
by sergey-safarov 28 May '25

28 May '25

#### Pre-Submission Checklist - [x] Commit message has the format required by CONTRIBUTING guide - [x] Commits are split per component (core, individual modules, libs, utils, ...) - [x] Each component has a single commit (if not, squash them into one commit) - [x] No commits to README files for modules (changes must be done to docbook files in `doc/` subfolder, the README file is autogenerated) #### Type Of Change - [x] Small bug fix (non-breaking change which fixes an issue) - [ ] New feature (non-breaking change which adds new functionality) - [ ] Breaking change (fix or feature that would change existing functionality) #### Checklist: - [x] PR should be backported to stable branches - [x] Tested changes locally - [ ] Related to issue #XXXX (replace XXXX with an open issue number) #### Description As a local IP address for TCP sending operation the Kamailio service is taking the same network_interface/IP_address, which is used by the service for TCP listening. You can view, comment on, or merge this pull request online at: https://github.com/kamailio/kamailio/pull/3925 -- Commit Summary -- * core: local TCP socket is bound on listening address -- File Changes -- M src/core/tcp_main.c (24) -- Patch Links -- https://github.com/kamailio/kamailio/pull/3925.patch https://github.com/kamailio/kamailio/pull/3925.diff -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/pull/3925 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/pull/3925(a)github.com>

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