- sr-users - kamailio.org

Time taken for tls_fix_domains_cfg initialization to complete is 8 seconds
by mahesh b 30 Apr '20

30 Apr '20

Hi All, I have 10 client profiles and 10 server profiles along with default client and server profile in my tls.cfg These are the numbers on time taken for tls_fix_domains_cfg initialization to complete based on the number of client and server profiles i have. Number of profiles Time diff (Sec) No of tcp child processes Start time End time 5 *3.99* 42 16:47:01.836 16:47:05.826 6 4.69 42 16:59:29.335 16:59:34.027 7 *5.347* 42 17:04:30.271 17:04:35.618 8 *6.10* 42 17:18:41.109 17:18:47.204 9 *6.614* 42 17:22:36.869 17:22:43.483 10 *7.379* 42 17:27:46.676 17:27:54.055 Kamailio 5.1.9 version is running on the below Linux Kernel version : 3.10.0-693.el7.x86_64 Centos version : CentOS Linux release 7.4.1708 (Core) CPU : 2 cores with model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [root@miv5000 ~]# cat /proc/meminfo MemTotal: 3882076 kB MemFree: 811244 kB MemAvailable: 2320356 kB Openssl verison : OpenSSL 1.0.2k-fips 26 Jan 2017 Now my question is : Is it ok for the api to take "8 seconds" to complete initialization in case of having 10 client and 10 server profiles along with default profiles in tls.cfg ? The time units taken for initialization to complete should it be in the order of milliseconds range or in seconds range ? Are there any more parameters related to tls configuration like "crl " which would play a factor in time taken for the initialization to complete. Currently at my application level , timeout is happening as the initialization is taking more time when more number of profiles are enabled. Please share your inputs on whether it is justifiable with my setup cpu/memory, it is ok for tls_fix_domains_cfg to take 8 seconds for the initialization to complete. Regards, Mahesh.B.

1 0

Next online devel meeting - Wed, Apr 29, 2020, 14:00UTC
by Daniel-Constantin Mierla 29 Apr '20

29 Apr '20

Hello, in order to plan the roadmap to next major release, it is time to organize another online devel meeting, so the developers and community members can syncronize and plan what should be done till Kamailio v5.4 will be out. Because freenode.net service is now requiring user registration to join IRC channels on their servers, we are considering some alternatives that would allow guest participation, one of them being using a matrix chat room. More details about it will be posted soon. I created a wiki page for the event at: * https://www.kamailio.org/wiki/devel/irc-meetings/2020a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, April 29, 2020, at 14:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 7

NAT and RTP
by Pavithra M 29 Apr '20

29 Apr '20

Hi, I have installed kamailio as a ims server in openstack and enabled WITH_NAT and rtpproxy . I have installed pcscf, icscf, scscf and hss in four different ubuntu instances . server - kamailio 5.3.3. When i make a call using zoiper client from external network , call is becoming a missed call without giving ACK and BYE. Openstack - pcscf - 10.40.10.3 (floating ip - 10.45.4.22) icscf - 10.40.10.12 scscf - 10.40.10.5 hss - 10.40.10.31 zoiper1 - 10.0.2.15 zoiper2 - 10.0.2.15 kindly help me in this regard. when i installed without using NAT and rtpproxy, the call went well. Since currently i am using Opensatck , it is my prerequisite to use zoiper client externally .so for me it is mandate to enable NAT and rtpproxy. Help me please!!!

1 0

change module parameter in routing logic
by Ali Taher 28 Apr '20

28 Apr '20

Hi, Is it possible to change a module parameter inside the routing logic ? For example I want to change sqlcon parameter of sqlops module based on the originating IP. Regards,

2 5

DNS SRV issue
by Melek Oktay 28 Apr '20

28 Apr '20

Hi All, I have a user, who has a SRV records like below, srv:betaproxy.siptelekom.com 1- SRV betaproxy siptelekom 4 hrs 20 0 5060 beta-proxy-dc01.siptelekom.com (X.Y.87.150) 2- SRV betaproxy siptelekom 4 hrs 10 0 5060 beta-proxy-dc02.siptelekom.com (X.Y.87.160) And I want to send the call to him/her, but one of the potential IP address could be failed. Assume that, first record (X.Y.87.150) is failed when I'm trying to send the packet. Later on, timeout occurs and our Kamailio try to send the packet to second one (X.Y.87.160) after period ot time. My question: Next time when I want to send another call to the same user, how Kamailio could use second (X.Y.87.160) SRV instead of trying to first one (X.Y.87.150) ? What is the elegant way to implement this ? Best Regards, Melek

1 0

RTP relay
by hugues 28 Apr '20

28 Apr '20

Hello I'm trying to run Kamailio in front of an asterisk server due to security issues. Both servers have a public ip address. It works fine but the RTP voice traffic goes directly from the VOIP client to the asterisk server without going through the kamailio server. I want to shut down the asterisk server so that it only talks to the kamailio server. UA -> Kamailio public IP addr <-> Asterisk public ip addr How do I force RTP traffic to go through kamailio. Apparently I don't have NAT so I don't need rtpproxy. But you need something that relays RTP traffic? If anyone has an idea I'll take it. Thank you for your help. Hugues. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

4 4

DMQ broadcasting crashes kamailio
by SamyGo 28 Apr '20

28 Apr '20

Hi, Is there a way to broadcast KDMQ to the cluster but not expect a reply back !?as far as I've read the source code dmq_bcast_message is exactly like dmq_send_message in a way that it expects a callback to be executed on response i.e expects a reply. So, the situation I'm facing is I'm broadcasting message to cluster and I do not want a reply back. The following two options result in crash & core dump. 1 - If my script doesn't respond back, by use of dmq_handle_message, it marks the destined servers as "inactive" and stops usrloc sync process which isn't desirable. 2 - If I respond back with the dmq_handle_message it crashes the Kamailio which just received this broadcasted message. Here is how its done in script: *broadcasting message to cluster:* dmq_bcast_message("userOnline", "$fu", "text/plain"); *Receiving and handling a broadcast message:* route[DMQ_HANDLE] { if(!(is_method("KDMQ") || $rm == "KDMQ")) return; if(is_method("KDMQ") || $rm == "KDMQ"){ if($rU =~ "userOnline"){ //user came online in cluster, resume transactions if-any suspended $avp(remoteUser) = $rb; } dmq_handle_message(); exit; } } *Related log lines:* Apr 23 21:15:48 kamailio[916]: ALERT: <script>: [da2c1-2f499] ------ DMQ_HANDLE: UserOnline Event Received ------ Apr 23 21:15:48 kamailio[916]: DEBUG: dmq [message.c:53]: ki_dmq_handle_message_rc(): dmq_handle_message [KDMQ sip:userOnline@9.8.7.123:5060] Apr 23 21:15:48 kamailio[916]: DEBUG: dmq [message.c:66]: ki_dmq_handle_message_rc(): dmq_handle_message peer found: userOnline Apr 23 21:15:48 kamailio[916]: DEBUG: <core> [core/receive.c:437]: receive_msg(): request-route executed in: 401461 usec Apr 23 21:15:48 kamailio[935]: DEBUG: dmq [worker.c:87]: worker_loop(): dmq_worker [2 935] lock acquired and crash/segfault.. Core dump: https://pastebin.com/S7ekCPfF Any help or pointers to solve this would be really appreciated. Best Regards, Sammy

5 13

RFC: C standard requirements
by Daniel-Constantin Mierla 27 Apr '20

27 Apr '20

Hello, (cross-posting because it is relevant to know where users need to run Kamailio as well as what developers tend to use for coding, but replies can go to one or the other list). Somehow I tried to stay close to ANSI-C/C90 with C coding style, for the reason of being compatible as much as possible with old compiles and niche/embedded systems. But I noticed that there are couple of C99 extensions in the core, like inline functions, variadic macros (the debugging macros), use of 'long long' or line comments starting with "//" (not so many, but still such comment style is used). Although I haven't searched for occurrences in core, but I remember seeing designated initializers (or initializing a structure by field names). These are part of C99, as listed by: * https://en.wikipedia.org/wiki/C99#Design Although likely GCC had some of them like custom extensions before the standard was out. On the other hand, newer standard C11 made some of those extensions optional: * https://en.wikipedia.org/wiki/C11_(C_standard_revision)#Optional_features So, my email here is to figure out if we should consider C99 as the requirement for adding C code or be more restrictive because there are still cases of usage on very specific systems that do not have such C compiler. We can eventually make a list of what is accepted or not from the recent versions of C standard to stay as much as possible portable across Unix'es. >From develeopers perspective, it would be good to learn if they use (in other projects, etc.) extensions of the new standards that they found useful and simplify their programming work. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

how to use get my domain name in VIA header (INVITE) instead of IP address.
by Nasida Yuriy 27 Apr '20

27 Apr '20

Hi guys, It sounds simple but I can not get domain name in VIA header (INVITE) instead of IP address. It is just FS ---INVITE --> Kamailio --INVITE--> endpoint Kamailio gets INVITE, and adds one more VIA with it's IP address. How to get domain name instead ? Please advice.

6 12

Multi INVITE issue
by Pavithra M 27 Apr '20

27 Apr '20

Hi, I have configured kamailio IMS in vm based images. I have configured pcscf,icscf,scscf,hss and ue in five different vms.I am able to successfully make a call via IMS. But when making a INVITE from alice to bob, it goes via ue -> pcscf pcscf -> scscf scscf -> icscf icscf -> scscf scscf -> pcscf (here multiple INVITE is happening and giving 483 Too Many Hops error and then the call is going to UE successfully) pcscf -> ue Can anybody help me what am i missing . why multi INVITE is happening there alone and giving me too many hops issue. Even after that issue how the call reaches the other UE. Kindly anyone help me in this regard atleast where to check this scenario.

2 1

Dialog module problem.
by Voip support 24 Apr '20

24 Apr '20

Hello, i am using dialog module in-memory to get all dialogs and show in a website. I use for this kamctl dialog show which is run quite often for autorefresh. I noticed that some calls get hung on kamailio while on caller side and callee side calls are disconnected. I captured a wireshark trace and for same caller IP i get many good calls as well some bad calls (not disconnected properly). When comparing the BYE packets i could not notice any difference. However in the "BAD" calls it looks like the BYE from caller get stuck on kamailio and is not sent/forwarded to callee side. I get BYE retransmissions from caller and all BYE requests got stuck on kamailio. After few seconds Callee side sends BYE and that BYE also get stuck. I am wondering how to trace this issue (differently than taking a wireshark trace) as i dont see any difference in the good BYE and the bad BYE. It looks like for any reason from time to time BYE is not forwarded to the other side and when the other side sents BYE its also not forwarded to call initiator. Please give me a clue how to continue on this. I am using dispatcher module to send the calls. Best regards Tom

2 2

nat_uac_test(8)
by Sergiu Pojoga 24 Apr '20

24 Apr '20

When receiving an INVITE over a specific LTE carrier, I'm seeing 'c=IN IP4 192.0.0.4' in SDP, which isn't technically a RFC1918 or RFC6598 IP address and thus nat_uac_test(8) fails. What elegant workaround can be done to catch such specific cases? Thanks.

5 11

Kamailio like SBC with Teams
by sip user 24 Apr '20

24 Apr '20

Hello good morning ... I am new to this list and I was starting to mess with Kamailio, mainly to set it up as SBC against Teams, in this case. But I can't get it to work for me. If I launch a call from the Teams, in the Kamailio I see: 1.- In syslog: Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 52.114.7.24 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 4160, type 3 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 171:1857:1187, 30 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa1f240, 23, 2, 0x7f90f2438f80), fd_no=17 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa1f240, 23, -1, 0x0) fd_no=18 called Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:4219]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 1 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:3902]: send2child(): selected tcp worker 1 10(23159) for activity on [tls:SBC_IP:5061], 0x7f90f2438f80 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1507]: handle_io(): received n=8 con=0x7f90f2438f80, fd=9 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSs<default> Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f90f2438f80 n=1468 fd=9 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= Apr 15 11:11:41 vps793907 kamailio[23122]: #026#003#003 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9c1700, 9, 2, 0x7f90f2438f80), fd_no=1 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f90f2438f80, FD 9 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1291]: tcp_read_req(): tcp_read_req: EOF Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9c1700, 9, -1, 0x10) fd_no=2 called Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7f90f2438f80, state -1, fd=9, id=30 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7f90f2432b40 Apr 15 11:11:42 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7f90f2438f80, -1 from 1 Apr 15 11:11:42 vps793907 kamailio[23122]: 13(23167) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7f90f2432b40 2.- With TCPDUMP: 11:13:09.311797 IP SBC_IP .1024 > SBC_IP .eu.sip-tls: Flags [S], seq 261244614, win 64240, options [mss 1440,nop,wscale 8,nop,nop,sackOK], length 0 11:13:09.311898 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [S.], seq 812357247, ack 261244615, win 29200, options [mss 1460,nop,nop,sackOK,nop,wscale 7], length 0 11:13:09.340358 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [.], ack 1, win 2053, length 0 11:13:09.340560 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [P.], seq 1:187, ack 1, win 2053, length 186 11:13:09.340578 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [.], ack 187, win 237, length 0 11:13:09.341361 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [P.], seq 1:1469, ack 187, win 237, length 1468 11:13:09.369606 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [.], ack 1469, win 2053, length 0 11:13:12.451498 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [S], seq 309084204, win 64240, options [mss 1440,nop,wscale 8,nop,nop,sackOK], length 0 11:13:12.451587 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [S.], seq 3275066862, ack 309084205, win 29200, options [mss 1460,nop,nop,sackOK,nop,wscale 7], length 0 11:13:12.707119 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1, win 2053, length 0 11:13:12.707311 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [P.], seq 1:187, ack 1, win 2053, length 186 11:13:12.707345 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [.], ack 187, win 237, length 0 11:13:12.708273 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [P.], seq 1:1469, ack 187, win 237, length 1468 11:13:12.965101 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1469, win 2053, length 0 11:13:12.965156 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [F.], seq 187, ack 1469, win 2053, length 0 11:13:12.965688 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [F.], seq 1469, ack 188, win 237, length 0 11:13:13.220871 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1470, win 2053, length 0 I can't "receive" anything. I have generated the certificates and configured in the tls.cfg of the Kamailio: [server:default] method = TLSv1.2 verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/ssl/cert.key certificate = /etc/letsencrypt/ssl/cert.crt ca_list = /etc/letsencrypt/ssl/ca.crt Within Kamailio itself I have it configured to return a 200 KeepAlive to Teams when it receives an OPTIONS: event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip: SBC_DNS :5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } And I have measured the record_route for this new one: record_route_preset(" SBC_DNS :5061;transport=tls", " SBC_IP :5061"); I have created a dispatcher.list: # setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional) 1:1:sip:sip.pstnhub.microsoft.com;transport=tls 0 3 socket=tls: SBC_IP :5061;ping_from= sip:SBC_DNS 2:1:sip:sip2.pstnhub.microsoft.com;transport=tls 0 2 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS 3:1:sip:sip3.pstnhub.microsoft.com;transport=tls 0 1 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS 4:1:sip:sip-all.pstnhub.microsoft.com;transport=tls 0 1 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS I think that one of the problems is that I do not send the OPTIONS to the Teams well, since it is on their panel, it indicates that the SBC is INACTIVE. I don't know if you could help me straighten this out a bit ... Thank you so much for everything.. a greeting

5 15

Check if Kamailio is going to CANCEL branch
by Ivan Ribakov 24 Apr '20

24 Apr '20

Hi all, Upon receiving a provisional response from callee in a branch route, is there a way of telling if Kamailio is going to automatically CANCEL that branch because a final reply was received from another branch before that? The closest I got is finding “cancel_b_method” parameter of the TM module, but that is not what I’m after. Previously I was recommended to debug via logging onreply_route but I’m not really sure what information may contain hints of intention to send CANCEL. Does anyone have any suggestions/guesses in which direction I should be digging? Thanks, Ivan

1 0

[info] matrix chat room for kamailio
by Daniel-Constantin Mierla 23 Apr '20

23 Apr '20

Thanks to the efforts made by Fred Posner, a matrix.org server has been installed on Kamailio project infrastructure (matrix.kamailio.dev - server sponsored by lod.com), which enables us to create chat rooms as we need for our community of users and developers. It tries to be an alternative to the existing IRC #kamailio channel, given that a while ago freenode.net had to enforce user registration to join the channels on their service (not complaining about it, they did it for good reasons to protect against spammers and D/DoS attacks). The #kamailio channel on freenode.net will still be in use by the people that like it, there is no plan to consider it obsolete. One of the main benefits of running own matrix server is the ability to allow guest access during the devel online meetings, so people that do not want to create an account can still participate by using a web browser. Also, it offers modern encryption and privacy, along with richer collaboration features (e.g., file sharing), many considering it a viable open source alternative to Slack or other private team collaboration services. The first room created is #kamailio:matrix.kamailio.dev : * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev The matrix server is federated, so you can join if you have an account on any other public federated matrix server or one that you can host yourself. The server on matrix.kamailio.dev is not allowing public registrations. If you are a registered developer (have git commit access) and need a matrix account, let us know and we will find solutions. It is the first phase now and based on the interest from the community, its usage can be increased (e.g., creating rooms for development notifications (commits, pull requests, issue); bridge to irc channel; etc...) or the server can be shut down if proves not useful. We will try to do the next online devel meeting (scheduled for Wednesday, April 29, 2020, at 14:00UTC) on the matrix chat room mentioned above. More details at: * https://www.kamailio.org/w/2020/04/matrix-chat-room-for-kamailio/ Stay safe and healthy! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio, TSILO and "local" calls
by German Cancio 23 Apr '20

23 Apr '20

Dear All, does somebody know up to which point TSILO can handle calls from one UA to another UA, when both are registered on the same Kamailio instance? Is a routing through a backend (such as Asterisk) required in that case? thanks and best regards, Germán

2 2

Integrating Push Notifications with iOS
by German Cancio 22 Apr '20

22 Apr '20

Dear All, we are working on the integration of VoIP Push Notifications with iOS devices via Kamailio. From what we observe, the time window between receiving a VoIP APNS notification that wakes up the client app and the client app being sent back to the background by iOS is extremely narrow. It is around 1-2 seconds, with a (seemingly random) variance of say ~0.5s. So 1-2s is the time window available for getting the client app to properly REGISTER and to receive the INVITE back from Kamailio (using e.g. TSILO); otherwise the call is lost. (On Android, the equivalent time window extends to 10s or more.) Have others observed the same on iOS? We are using a client app that still uses the iOS 12 SDK (with Xcode10). Can we expect changes in that regard with iOS13/Xcode13? many thanks and cheers, Germán

3 5

doubt about xflags
by David Escartin 22 Apr '20

22 Apr '20

Dear all one quick question, reading the module corex doc, seems that xflag are message(transaction) flags. But I made a test and seems for some reason the flag is not seeing activated at the onreply_route, when it's activated on the request route. Seemed more like a script flag behaviour. Maybe I'm missing something? thanks a lot and regards david

2 8

Parallel forking - first responder wins
by Ivan Ribakov 21 Apr '20

21 Apr '20

Hi all, What I’m trying to achieve is to have Kamailio fork an INVITE to multiple endpoints in parallel but only maintain the branch that responds first (first to respond with 200 OK I guess). I’ve read the TM module documentation on forking (https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.serial_fork… <https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.serial_fork…>) and as far as I understood a combination of “seturi()” + “append_branch()” + “t_relay()” command calls will allow me to send multiple forked INVITEs in parallel. What I couldn't find information about in the documentation (please point me to it in case I missed it) is what controls (if any) do I have over forked requests. Do I need to keep track of the branches myself and cancel others when first succeeds or does Kamailio have some kind of setting for implementing such behaviour? Thanks, Ivan

7 12

NAT pinging after service restart
by Jacob Greene 21 Apr '20

21 Apr '20

Hello everyone! When I restart the Kamailio service, NAT pinging for all existing location records stops and does not start until the UA registers again. I'm using db_mode 2, and all of the registrations will be there, but for some reason Kamailio wont restart the pinging cycle. Is it possible to configure Kamailio to restart NAT pinging in the event of a service restart? Thanks for reading! Jake Greene Voice Engineer

1 1

Join us for SIP Chronicles Live #1, featuring Giovanni Maruzelli
by Maxim Sobolev 20 Apr '20

20 Apr '20

Dear Real-Time Friends and Colleagues! As many of you we have been totally devastated that we will have no chance to see you in the next few months to come. :-/ Some people in the community believe it might be years. I don’t necessarily agree with that opinion myself. Over the course of the last few years our team had a great time extending live coverage for some of those events that have been affected, got some experience and equipment. Instead of just waiting for the virus to clear, we decided to organize a series of bi-weekly live casts with some of the speakers that we have hoped to see at those events presenting their latest developments live and then answering questions from the audience. So without further ado, let me introduce our first guest Giovanni Maruzzelli, who is going to introduce his newest project SaraPhone ( https://github.com/gmaruzz/saraphone). Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask Giovanni a question about his project live: https://youtu.be/mF9elIcVGE8 Or if you miss that opportunity, you can always watch the recording later on Sippy Labs channel on YouTube and email Giovanni your question at < gmaruzz(a)gmail.com>. SaraPhone is a bare bone SIP WebRTC phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) Stay healthy, optimistic and productive! Also share, like and subscribe. See you soon!!! Regards, Max

1 0

INVITE forking - uniform way to identify branches
by Ivan Ribakov 20 Apr '20

20 Apr '20

Hi all, I have a prioritised list of endpoints, where endpoints with the same priority should be sent in parallel: - endpoint_1_uri; p = 1 - endpoint_2_uri; p = 2 - endpoint_3_uri; p = 4 - endpoint_4_uri; p = 5 As some of the lower priority endpoints are not known upfront, I can not load entire list into the contact list and use TM module's “forking based on Q value” to take care of the rest. So my plan is to load all endpoints with the same priority and fork them in parallel. If none succeed - obtain next endpoint(s) and repeat the process. As I go through the above process, I need to update some counters each time delivery to one of the endpoints succeeds or fails. To be able to perform these “housekeeping” tasks I need to be able to clearly identify branches and link them to the destination URI (endpoint URI) of the forked INVITE that started them. Ideally I would like to be able to maintain relationship “branch_id” => “destination URI” where “branch_id” is the identifier that is appended to Via branch parameter. What is the best/most reliable way to achieve this? I tried logging and observing the value of $du - it was available just before sending out a forked INVITE, but in branch_failure and onreply_route it was <null>. I tried logging and observing $T_req($du) - I only expected it to work from onreply_route, but it was also <null>. I tried logging and observing $T_branch_idx - while it was showing meaningful data before sending forked INVITE and during branch_failure route, I couldn’t access it from onreply_route so I couldn’t tell which of the branches current reply is for. Are there any more variables that could help me build “branch_id” => “destination URI” relationship and access information about “branch_id” and/or “destination URI” from branch_failure and onreply_route? Apologies in advance for a lengthy description, I did my best to summarise the situation and what I’m trying to achieve. Regards, Ivan

1 0

Re: [SR-Users] (no subject)
by ahmed moghazy 20 Apr '20

20 Apr '20

Excuse me sir, if i want to make integration between kamailio and asterisk to make kamailio could use land line , i followed steps but i got errors i want help , was asterisk is only way to make kamailio use landline (fxs,fxo) Sent from my iPhone > On Mar 10, 2020, at 9:52 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

2 1

LCR and fallback
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi! I would like to implement LCR and fallback capabilities for my Kamailio installation. I'm currently using the dispatcher module with a simple configuration file. As I understood, there are multiple modules (carrier route, dynamic routing) that can be used to achieve that, what I'm aiming for is - * Country-based (or prefix-based) routing with up to 4 different carriers (could be expanded in the future, but I won't have more than a dozen), while also maintaining a fallback route if the 1st priority is not available. * Each of my carriers have his own requirements for headers and formatting, for instance, for one of our carriers, having a '+' prefix before dialing to a non-US number is mandatory, while on the others - it is not, or another example - one of my carriers doesn't allow having a P-Asserted-Identity header in my invite, while others do. I would love to have the option to manipulate headers based on the route that I have chosen, both on the default and fallback routes. * It would be nice to receive SMTP or SNMP alerts if a fallback does occur. * Easy management, having the ability to load and reload the routing table on a monthly basis, once the pricing list is changed. What would be your suggestion? Edward

2 1

How to install the PHONENUM module?
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi, I'm trying to add the PHONENUM module into my Kamailio installation. For this, I've cloned the main branch of Libphonenumber<https://github.com/google/libphonenumber/tree/master/cpp> and followed the installation rules as they appear in the relevant README file: Building and testing the library -------------------------------- $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test It all went through and the library was created, but when I try to make and install the module itself, I'm getting the following error. Looks like the library wasn't included in the building context, can you please lend me a hand and tell me how do I include it? root@kamaillioegress:/usr/local/src/kamailio-5.1/kamailio/src/modules/phonenum# make Compiling cphonenumber.cpp g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -DNAME='"kamailio"' -DVERSION='"5.2.0-dev6"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 5.4.0"' -D__CPU_x86_64 -D__OS_linux -DVERSIONVAL=5002000 -DCFG_DIR='"/usr/local/kamailio-5.1/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DMEM_JOIN_FREE -DF_MALLOC -DQ_MALLOC -DTLSF_MALLOC -DDBG_SR_MEMORY -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_IP_MREQN -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/opt/local/include -DKAMAILIO_MOD_INTERFACE -DMOD_NAME='"phonenum"' -c cphonenumber.cpp -o cphonenumber.o cphonenumber.cpp:24:65: fatal error: phonenumbers/geocoding/phonenumber_offline_geocoder.h: No such file or directory compilation terminated. Makefile:22: recipe for target 'cphonenumber.o' failed make: *** [cphonenumber.o] Error 1 Edward

2 1

Free Persian Asterisk Book and COVID-19 Pandemic
by Mojtaba 19 Apr '20

19 Apr '20

In the context of the COVID-19 pandemic, Nasim Telecom decided to publish the first six-captures of Persian Asterisk book that was written by me and Mr Najafi, published in 2017. Also it was introduced by Mr. David Duffett in AstriCon 2017 on Florida. It would be useful for all people which understand Persian language, Also all other people could use it by going code commands on it and examples. For downloading it go the following link: https://www.nasimtelecom.com/en/persian-asterisk-book/ So, be careful and safety at first, Stay at home and read your favourite books. With Best Regards. --Mojtaba Esfandiari.S

3 2

rtpenigne & STUN
by Ahed Saed 17 Apr '20

17 Apr '20

Hi, I'm using kamailio 5.3 & rtpengine ( both kamailio and rtpengine resides on public IP on same machine) am using below logic to handle clients behind NAT if(nat_uac_test("8")){ rtpengine_manage("SIP-source-address replace-origin replace-session-connection "); }else{ rtpengine_manage("trust-address replace-origin replace-session-connection "); } I don't want all the rtp packet to go over the server, i just want to make it use the rtpengine just in case it is needed. but am not sure how this can be done using kamailio, from the client side i'll try to use STUN, but i understand STUN doesn't work all the time ( ~90% of the cases it will work) can anyone help me with this ? another question - if one client works with STUN and the other is behind NAT how this should be handled? Cheers

1 0

TCP/TLS connection (id: 0) for WebSocket could not be found
by Mack Hendricks 17 Apr '20

17 Apr '20

Hey All, I get this message when trying to route request to a WebSocket client: Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:1762]: check_boundaries(): no multi-part body Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:429]: clen_builder(): content-length: 651 (651) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: WARNING: <core> [core/msg_translator.c:2786]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: <core> [core/msg_translator.c:2002]: build_req_buf_from_sip_req(): could not create Via header Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -2 (-2) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:355]: t_relay_to(): -2 error reply generation delayed Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1176 a=24 n=sl_reply_error Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1178 a=2 n=exit Here is what my location table looks like. It should try to send the call to sip:98.209.240.245:56291;transport=ws *************************** 2. row *************************** id: 124 ruid: uloc-5e9685e8-5650-92 username: 2000 domain: ws-test.com contact: sips:2000@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss received: sip:98.209.240.245:56291;transport=ws path: NULL expires: 2020-04-15 05:03:17 q: -1.00 callid: 7ef0a550-b2b9-24c6-5c3b-8e0c667f5533 cseq: 50096 last_modified: 2020-04-15 04:59:57 flags: 0 cflags: 524352 user_agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 socket: NULL methods: NULL instance: NULL reg_id: 0 server_id: 0 connection_id: -1 keepalive: 1 partition: 18 The Websocket list looks like this: Server01:~# kamcmd ws.dump { connections: { 8: wss:98.209.240.245:56291 -> wss:134.122.27.49:4443 (state: OPEN, last used 47s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } I’m not sure what I’m doing wrong. Any ideas? Thanks in advance -Mack

2 8

Kamailio in osm
by Pavithra M 17 Apr '20

17 Apr '20

Hi, Do anybody have worked orchestrating kamailio as an IMS server through osm by launching vnfs in openstack . If so kindly reply me as I need some inputs on that.

1 0

Dealing with mobile client state while using Push Notification and TSILO
by Hubert Odziemczyk 17 Apr '20

17 Apr '20

Hello! We are trying to setup a call-flow with TSILO and Push Notifications for solution with a mobile SIP client (Linphone), based on presentation by Federico Cabiddu (http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…). We have an issue with "managing" state of the client (active or not) when there is a call coming. To be more specific, lookup in location table doesn't always give a proper answer, as contact expiration is different from actual lifetime of the app (which also differs between iOS and Android). One of proposed solutions is setting Expires=1 for REGISTER and then always rely on Push Notifications, but this scenario seems to be unreliable with some clients that are in foreground, as Push Notification doesn't trigger REGISTER again. Question is: did you perceive similar problems? What are your solutions to deal with them? Thank you in advance! Best regards, Hubert

2 1

Announcing SaraPhone, SIP WebRTC Open Source business phone
by Giovanni Maruzzelli 16 Apr '20

16 Apr '20

My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, Kamailip, OpenSIPS, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

kamailio integration with MS teams.
by Nasida Yuriy 16 Apr '20

16 Apr '20

Hi guys, Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks. Regards, Team lead<https://www.odesk.com/companies/Nasida_~0168bc9fd46660bcca> [https://odesk-prod-portraits.s3.amazonaws.com/Companies:1712760:CompanyLogo…]

4 6

Weird dialplan result
by David Villasmil 15 Apr '20

15 Apr '20

Hello guys, So i have the following dialplan (almost literally, 2 records) +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | 10 | 1 | 100 | 0 | 910000000 | 0 | | | 1.2.3.4:5060 | | 11 | 1 | 100 | 0 | 910000001 | 0 | | | sip.somedomain.net:5060 | And with kamcmd: kamcmd> dialplan.translate 1 s:910000000 { Output: Attributes: 1.2.3.4:5060 } and kamcmd> dialplan.translate 1 s:910600000 error: 500 - No translation So that works... now, when i get an INVITE I do: dp_match("1", "$rU"); xlog("[DIALPLAN] '$rU' was translated to '$var(destination)'\n"); if ( $var(destination)!=0 ) { $ru = "sip:" + $fU + "@" + $var(destination); route(RELAY); } And this prints: <script>: [DIALPLAN] '910600000' was translated to '195.114.216.188:5060' What am I doing wrong? Kind of confused here... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 15 Apr '20

15 Apr '20

Good day, I am testing the dispatcher module, using Kamailio as stateless proxy. I have a pool of UAC (scripts in SIPP) and a pool of UAS (also scripts in SIPP) for the destinations. Kamailio version is kamailio-5.3.3-4.1.x86_64. Problem I have is, if UAS responds 180 and 200 OK to Invite immediately, sometimes they are propagated out of order. 200 OK before 180, like this : UAS is 172.30.4.195:5061. UAC is 172.30.4.195:5080. Kamailio is 192.168.253.4:5070 Difference between 180 and 200 is just about 50 microseconds. My guess is that both messages are received by different instances of Kamailio, and then because of context switches, even though the 180 is received before, that process ends after the processing of 200. However, I had the idea that in order to avoid these problems the kamailio processes synchronized with each other using a shared memory. I tried using stateful proxy and I obtained the same result. By the way, anyone has any idea about how Kamailio's share memory is implemented? It clearly does not use the typical system calls shmget(), shmat(), because they are not shown by ipcs command. Before posting here I googled, but I couldn't find anything related to this. I can't believe I am the only one who ever had this problem, so I guess I am doing something wrong... Please, any help. I'm really stuck on this. Thanks. -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

6 11

Re: [SR-Users] [EXT] UE client for IMS
by Woscek, Martin W. 15 Apr '20

15 Apr '20

https://code.google.com/archive/p/boghe/downloads Starting with version labeled: Boghe_2.0.113.744.sfx.exe<https://storage.googleapis.com/google-code-archive-downloads/v2/code.google…> works with IPv6. From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Pavithra M Sent: Tuesday, April 14, 2020 4:28 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] [SR-Users] UE client for IMS Hi, Could anyone suggest me what UE client can be used to test the normal ims call flow . Currently I am using zoiper client and facing issues of 403 forbidden - domain not served . So kindly help me in this regard so that if I change my UE client it works . Thanks

1 0

UE client for IMS
by Pavithra M 15 Apr '20

15 Apr '20

Hi, Could anyone suggest me what UE client can be used to test the normal ims call flow . Currently I am using zoiper client and facing issues of *403 forbidden - domain not served *. So kindly help me in this regard so that if I change my UE client it works . Thanks

2 1

kamailio integration with MS teams.
by Yuriy Nasida 14 Apr '20

14 Apr '20

Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks.

1 0

H264 codec licence
by Prakash Ganesan 14 Apr '20

14 Apr '20

Hi, We are planning to use kamilio/RTP engine for our WebRTC calls. I would like to know how the H264 codec licences are handled as i know there is a royalty fee associated with this video codec from the server perspective. As of now there is plan to use H264 pass through and vp8/vp9 transcoded to H264 by the kamilio/RTP engine. -- Cheers Prakash

2 1

Using dispatcher module : Modifying message before dispatching with send_udp()
by Luis Rojas G. 13 Apr '20

13 Apr '20

Hello, I am using the dispatcher module, Kamailio version 5.3.3. The point is, I am using send_udp() to forward the message, because I don't want Kamailio to add headers Via nor Record-Route. I am just dispatching the Invite message, and the rest of communication to be made just between the endpoints, not through Kamailio. However, I need to add before a custom header, using append_hf(). It seems, send_udp() really changes nothing at all in the message, not even something added by the script with append_hf(). It works if I use forward(), but not if I use send_udp(). Is there a way to accomplish this? Best regards, -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

2 2

Lifetime of Invite with no Ack
by Duarte Rocha 13 Apr '20

13 Apr '20

Greetings, How much time will Kamailio keep a transaction active if the ACK to a INVITE is never received by Kamailio? Best Regards,

3 7

Using Stir/Shaken with Kamailio
by Alexandru Tripon 13 Apr '20

13 Apr '20

Hi, After I watched the talk about Stir/Shaken(https://vimeo.com/396940577), I wanted to try the module(secsipid), but I couldn't make it work. I built the Kamailio from source using the following steps: https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . And then built the secsipid following the instruction found here: https://www.kamailio.org/docs/modules/devel/modules/secsipid.html#idm90 . But when I start the Kamailio with the module loaded I get the following error: ` kamailio: ERROR: <core> [core/sr_module.c:325]: version_control(): no version info in module </usr/local/lib64/kamailio/modules/secsipid.so>: /usr/local/lib64/kamailio/modules/secsipid.so: undefined symbol: module_version ` How can I fix this? Thank you in advance, Alexandru Tripon

2 2

Callee BYE routing issue
by Daniel W. Graham 13 Apr '20

13 Apr '20

Have an issue with BYE routing in a lab setup, any suggestions on what I should be looking at? Topology: UA1 <-> NAT device 1 <-> Proxy1 <-> Registrar/B2BUA <-> Proxy1 <-> NAT device 1 <-> UA2 All UA to UA calls flow through B2BUA. All IP’s are public except for UA1 and UA2. -Both endpoints can call each other. -If the caller hangs up first, BYE is routed properly. -If the callee hangs up first, BYE causes proxy to send 478. On calls where BYE results in 478: if(!isdsturiset()) { ……….…………………….not executed handle_ruri_alias() } return; Example issue call: UA2 to UA1, UA1 hang-up. Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=16 n=if Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=532 a=24 n=isdsturiset Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=2 n=return ……….. Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} tm [ut.h:245]: uri2dst2(): bad_uri: [sip::;transport=] Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} tm [t_fwd.c:1732]: t_forward_nonack(): failure to add branches Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=275 a=24 n=sl_reply_error Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} sl [sl_funcs.c:401]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=277 a=2 n=exit 2020/04/10 00:31:28.625060 NAT_Device_IP:57808 -> Proxy_IP:5060 BYE sip:398@B2BUA_IP:5160 SIP/2.0 Via: SIP/2.0/UDP UA1_IP:52884;branch=z9hG4bK-524287-1---43b5f31829f20952 Max-Forwards: 70 Route: <sip:Proxy_IP;lr=on;ftag=as78671fd9;did=b1b.a27> Contact: <sip:399@UA1_IP:52884> To: "Dan Test" <sip:398@B2BUA_IP:5160>;tag=as78671fd9 From: <sip:399@UA1_IP:52884;rinstance=705e0af5f4be485c;alias=NAT_Device_IP:~57808~1>;tag=64f4b712 Call-ID: 24dc84a4592515245177483c2a657042@B2BUA_IP:5160 CSeq: 2 BYE User-Agent: xxxxxxxxxxxxxxxx Content-Length: 0 ######################################### For comparison, here is a BYE from another test call: UA1 to UA2, UA1 hang-up. This BYE was routed correctly. 2020/04/10 00:44:02.688608 NAT_Device_IP:57808 -> Proxy_IP:5060 BYE sip:398@B2BUA_IP:5160 SIP/2.0 Via: SIP/2.0/UDP UA1_IP:52884;branch=z9hG4bK-524287-1---ce36df02e8a80a48 Max-Forwards: 70 Route: <sip:Proxy_IP;lr=on;ftag=9078f232;did=49e.8c01> Contact: <sip:399@UA1_IP:52884> To: <sip:398@B2BUA_IP>;tag=as5874901f From: <sip:399@B2BUA_IP:5060>;tag=9078f232 Call-ID: 103104NGU1NjkxNGZkYmM4MzA3Y2FkMzY1OGNkZTZmMTMyZDU CSeq: 3 BYE User-Agent: xxxxxxxxxxxxxxxx Authorization: Digest username= xxxxxxxxxxxxxxxx Content-Length: 0 -dan

4 7

Kamailio as IMS (version - 5.3.3) Error : 403 Forbidden Domain not served
by Pavithra M 13 Apr '20

13 Apr '20

Hi David, Kindly help me in this regard. I am not getting like it is because of pval issue or some other issue which causes this issue. I have configured kamailio as an IMS server as VM based . PCSCF - 10.x.x.179 ICSCF - 10.x.x.79 SCSCF - 10.x.x.73 HSS - 10.x.x.75 I am using zoiper client for testing the call flow . The *registration is happening properly* in my both zoiper clients for hss configured users (alice and bob). When i am making an *INVITE , *I am getting *403 Forbidden - Domain not served *error from SCSCF. The Call Flow for invite is as follows . call from alice(a)sip.example.com zoiper1 (UE) -> PCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) PCSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> PCSCF (100 trying) SCSCF -> ICSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) ICSCF -> SCSCF (100 trying) ICSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> ICSCF (100 trying) SCSCF -> ICSCF (*403 Forbidden - Domain not served*) ICSCF -> SCSCF (Request: ACK sip:bob@sip.example.com | ) ICSCF -> SCSCF ( 403 Forbidden - Domain not served ) SCSCF -> PCSCF ( 403 Forbidden - Domain not served ) PCSCF -> UE ( 403 Forbidden - Domain not served ) Below are the alias configuration i have done in pcscf,icscf,scscf nodes *PCSCF.CFG* # IP-Adress for incoming SIP-Traffic, in the following format: # SIP / UDP listen=udp:10.x.x.179:4060 # SIP / TCP listen=tcp:10.x.x.179:4060 # SIP / TCP/TLS listen=tls:10.x.x.179:4060 # IPSEC / UDP #!define IPSEC_LISTEN_ADDR "0.0.0.0" #!define IPSEC_CLIENT_PORT 5062 #!define IPSEC_SERVER_PORT 5063 #alias="sip.example.com:4060" alias=pcscf.sip.example.com:4060 #!define MY_WS_PORT 80 #!define MY_WSS_PORT 443 #!define PCSCF_URL "sip:pcscf.sip.example.com:4060" #!define TCP_PROCESSES 8 #!subst "/NETWORKNAME/sip.example.com/" #!subst "/HOSTNAME/pcscf.sip.example.com/" *ICSCF.CFG* # SIP / UDP listen=udp:10.x.x.79:4070 # SIP / TCP listen=tcp:10.x.x.79:4070 # SIP / TCP/TLS listen=tls:10.x.x.79:4071 alias="icscf.sip.example.com:4070" #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "icscf.sip.example.com" *SCSCF.CFG* # SIP / UDP listen=udp:10.x.x.73:4080 # SIP / TCP listen=tcp:10.x.x.73:4080 # SIP / TCP/TLS listen=tls:10.x.x.73:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 # ENUM-Server to query: #!define ENUM_SUFFIX "scscf.sip.example.com." Kindly let me know where i am missing . Kindly need help in this issue since the registration is happening successfully and dropping in the INVITE level. Kindly help.

1 0

kamailio error
by Pavithra M 13 Apr '20

13 Apr '20

Hi, This is the lvalue.c file where i am facing the issue. kindly help. if (unlikely(ret<0)){ LM_ERR("assignment failed at pos: (%d,%d-%d,%d)\n", rve->fpos.s_line, rve->fpos.s_col, rve->fpos.e_line, rve->fpos.e_col); } else { if(unlikely(_log_assign_action!=NULL)) _log_assign_action(msg, lv); } rval_destroy(rv); return ret; error: if (rv) rval_destroy(rv); return -1;

1 0

Kamailio - 5.3.3 Error : 403 domain not served
by Pavithra M 13 Apr '20

13 Apr '20

Hi , I have configured kamailio as an IMS server as VM based . PCSCF - 10.x.x.179 ICSCF - 10.x.x.79 SCSCF - 10.x.x.73 HSS - 10.x.x.75 I am using zoiper client for testing the call flow . The *registration is happening properly* in my both zoiper clients for hss configured users (alice and bob). When i am making an *INVITE , *I am getting *403 Forbidden - Domain not served *error from SCSCF. The Call Flow for invite is as follows . call from alice(a)sip.example.com zoiper1 (UE) -> PCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) PCSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> PCSCF (100 trying) SCSCF -> ICSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) ICSCF -> SCSCF (100 trying) ICSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> ICSCF (100 trying) SCSCF -> ICSCF (*403 Forbidden - Domain not served*) ICSCF -> SCSCF (Request: ACK sip:bob@sip.example.com | ) ICSCF -> SCSCF ( 403 Forbidden - Domain not served ) SCSCF -> PCSCF ( 403 Forbidden - Domain not served ) PCSCF -> UE ( 403 Forbidden - Domain not served ) Below are the alias configuration i have done in pcscf,icscf,scscf nodes *PCSCF.CFG* # IP-Adress for incoming SIP-Traffic, in the following format: # SIP / UDP listen=udp:10.x.x.179:4060 # SIP / TCP listen=tcp:10.x.x.179:4060 # SIP / TCP/TLS listen=tls:10.x.x.179:4060 # IPSEC / UDP #!define IPSEC_LISTEN_ADDR "0.0.0.0" #!define IPSEC_CLIENT_PORT 5062 #!define IPSEC_SERVER_PORT 5063 #alias="sip.example.com:4060" alias=pcscf.sip.example.com:4060 #!define MY_WS_PORT 80 #!define MY_WSS_PORT 443 #!define PCSCF_URL "sip:pcscf.sip.example.com:4060" #!define TCP_PROCESSES 8 #!subst "/NETWORKNAME/sip.example.com/" #!subst "/HOSTNAME/pcscf.sip.example.com/" *ICSCF.CFG* # SIP / UDP listen=udp:10.x.x.79:4070 # SIP / TCP listen=tcp:10.x.x.79:4070 # SIP / TCP/TLS listen=tls:10.x.x.79:4071 alias="icscf.sip.example.com:4070" #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "icscf.sip.example.com" *SCSCF.CFG* # SIP / UDP listen=udp:10.x.x.73:4080 # SIP / TCP listen=tcp:10.x.x.73:4080 # SIP / TCP/TLS listen=tls:10.x.x.73:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 # ENUM-Server to query: #!define ENUM_SUFFIX "scscf.sip.example.com." Kindly let me know where i am missing . Kindly need help in this issue since the registration is happening successfully and dropping in the INVITE level. Kindly help.

1 0

Kamailio 5.3.2 crashing continuously
by harneet singh 13 Apr '20

13 Apr '20

Hi All, As part of our very low cps load at 50 calls per second through Kamailio Version 5.3.2 (Dispatcher Module used with Call Load based Routing - Algorithm 10. Machine is a Centos: 7.7 with kernel version 3.10), we are seeing the kamailio continuously crashing. Please see the data below for 3 types of segfaults seen: Crash-1: -------- Apr 3 19:32:25 FE-A07-34-VM6 kernel: kamailio[10382]: segfault at 7ff300000078 ip 00007ff300000078 sp 00007ffe5f2eafc8 error 14 in libbz2.so.1.0.6[7ff331582000+f000] Apr 3 19:32:25 FE-A07-34-VM6 mysqld: 2020-04-03 19:32:25 6131 [Warning] Aborted connection 6131 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10423]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10381]: ALERT: <core> [main.c:767]: handle_sigs(): child process 10382 exited by a signal 11 Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10381]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated =============================== Crash-2: ------- Apr 3 19:36:11 FE-A07-34-VM6 kernel: kamailio[12838]: segfault at 7efe00000078 ip 00007efe00000078 sp 00007ffdc876b868 error 15 in zero (deleted)[7efdc407f000+40000000] Apr 3 19:36:11 FE-A07-34-VM6 mysqld: 2020-04-03 19:36:11 6175 [Warning] Aborted connection 6175 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12874]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 18 Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12826]: ALERT: <core> [main.c:767]: handle_sigs(): child process 12838 exited by a signal 11 Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12826]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated ============================ Crash-3: -------- Apr 3 19:40:53 FE-A07-34-VM6 kernel: kamailio[13542]: segfault at 80 ip 000000000065e193 sp 00007ffdae10c1f0 error 4 in kamailio[400000+476000] Apr 3 19:40:53 FE-A07-34-VM6 mysqld: 2020-04-03 19:40:53 6222 [Warning] Aborted connection 6222 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13516]: ALERT: <core> [main.c:767]: handle_sigs(): child process 13542 exited by a signal 11 Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13558]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 38 Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13516]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated Any pointers for resolution are most welcome as we need to quickly resolve this. Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

IPSec and USIM
by Mohammad Elahi 12 Apr '20

12 Apr '20

Hi All, When my VoLTE Mobile is registered to the Kamailio IMS that support IPSec by AKAv1 algorithm, inside the ESP packet is clear text (not encrypted). Is that because my mistake in configuration or for my USIM card? thanks in advance. Mohammad

1 1

kamailio as SBC for Teams
by Aidar Kamalov 10 Apr '20

10 Apr '20

Hello, I read https://skalatan.de/en/blog/kamailio-sbc-teams and trying to use kamailio for teams. But I faced that MS not responding ACK to my 200OK. Somethin is wrong, but I don't know what. I call from teams to kamailio. Can you help me please This is initial INVITE from Teams: 2020/04/09 22:55:17.346092 52.114.7.24:1024 -> KAMAILIO_IP:5061 INVITE sip:+xxxxxxxxx@MY_KAMAILIO_FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: Aidar Kamalov<sip:+xxxxxxxxx@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=de28a5b5333b451a97312097f91f5564 TO: <sip:xxxxxxxx@MY_KAMAILIO_FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: 3d5ad3cdd01457be94d127fa20478cf1 MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.7.24:5061;branch=z9hG4bKdb903e2c RECORD-ROUTE: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443 ;x-i=903eeaac-b19e-43a0-8e75-bb222049922a;x-c=3d5ad3cdd01457be94d127fa20478cf1/d/8/d3d2c2e3f3c14f88bb2fe843b4bc70e4> CONTENT-LENGTH: 2426 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.4.3 i.ASEA.5 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY v=0 o=- 0 0 IN IP4 52.114.252.146 s=session c=IN IP4 52.114.252.146 b=CT:50000 t=0 0 m=audio 52939 RTP/SAVP 104 9 111 18 0 8 103 97 13 118 119 101 a=rtcp:52925 a=ice-ufrag:rKBm a=ice-pwd:BkjUnkfIvYV5pjteSteevL1O a=rtcp-mux a=candidate:4 1 UDP 184547839 52.114.252.146 52939 typ relay raddr 31.13.144.55 rport 57780 a=candidate:4 2 UDP 184547326 52.114.252.146 52925 typ relay raddr 31.13.144.55 rport 57781 a=candidate:1 1 tcp-act 2121006590 100.78.247.179 1024 typ host a=candidate:1 2 tcp-act 2121006590 100.78.247.179 1024 typ host a=candidate:2 1 UDP 2130705919 100.78.247.179 39182 typ host a=candidate:2 2 UDP 2130705406 100.78.247.179 39183 typ host a=candidate:3 1 UDP 1694497791 31.13.144.55 57780 typ srflx raddr 100.78.247.179 rport 39182 a=candidate:3 2 UDP 1694497278 31.13.144.55 57781 typ srflx raddr 100.78.247.179 rport 39183 a=candidate:5 1 tcp-act 1684796926 31.13.144.55 9761 typ srflx raddr 100.78.247.179 rport 41775 a=candidate:5 2 tcp-act 1684796926 31.13.144.55 9761 typ srflx raddr 100.78.247.179 rport 41775 a=candidate:6 1 tcp-pass 174453758 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:6 2 tcp-pass 174453758 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:7 1 tcp-act 174846462 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:7 2 tcp-act 174846462 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=x-candidate-info:4 network-type=WWAN a=x-candidate-info:1 network-type=WWAN a=x-candidate-info:2 network-type=WWAN a=x-candidate-info:3 network-type=WWAN a=x-candidate-info:5 network-type=WWAN a=x-candidate-info:6 network-type=WWAN a=x-candidate-info:7 network-type=WWAN a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31 a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31|1:1 a=rtpmap:104 SILK/16000 a=fmtp:104 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 SILK/8000 a=fmtp:103 useinbandfec=1; usedtx=0 a=rtpmap:97 RED/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:119 CN/24000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:200 and my 200OK to MS is: 2020/04/09 22:55:19.976427 KAMAILIO_IP:5061 -> 52.114.7.24:1024 SIP/2.0 200 OK Record-Route: <sip:KAMAILIO_FQDN_DOMAIN:5061;transport=tls;ftag=de28a5b5333b451a97312097f91f5564;lr=on> Record-Route: <sip:KAMAILIO_IP:5060;ftag=de28a5b5333b451a97312097f91f5564;lr=on> Via: SIP/2.0/TLS 52.114.7.24:5061;rport=1024;branch=z9hG4bKdb903e2c Record-Route: <sip:ENDPOINT_BEHIND_KAMAILIO_IP;line=sr-QyfKA15Kw1rKw1hKEoPljM0oE1VLX10RQ1VLX10gkY37WJufE1BBjJhVE6E6r14VEJ5Nj6ElE1UNjyrNEJrVjnr.AyudmMV1rg52rJrlAyNBkMVNmSE*> Record-Route: <sip:ENDPOINT_BEHIND_KAMAILIO_IP;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.af1;nat=yes> Record-Route: <sip:KAMAILIO_IP;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.42a1;nat=yes> Record-Route: <sip:KAMAILIO_IP:5061;transport=tls;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.42a1;nat=yes> Record-Route: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> To: <sip:+xxxxxxxxxx@KAMAILIO_FQDN_DOMAIN :5061;user=phone>;tag=3795432917-393759697 From: <sip:+xxxxxxxxxx@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=de28a5b5333b451a97312097f91f5564 Call-ID: 3d5ad3cdd01457be94d127fa20478cf1 CSeq: 1 INVITE Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL Content-Type: application/sdp Accept: application/sdp Content-Length: 282 User-Agent: MY_DOMAIN Contact: <sip:KAMAILIO_IP;line=sr-DHZToOtmMdn7MODy9AnmD10TGLpUrL4noJCE4A03DdQE9LUaMXr*> v=0 o=ETPI-MSX1 0 0 IN IP4 KAMAILIO_IP s=sip call c=IN IP4 KAMAILIO_IP t=0 0 m=audio 12176 RTP/SAVP 8 b=AS:64 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtcp:12177 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:03ZmcR+CxxB7k7C8QvWBgseYq7cH7py7VxqfLM8j a=ptime:20 -- Aydar A. Kamalov

4 8

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 10 Apr '20

10 Apr '20

Hello, I just realized that I had the dispatcher configured using a hash of Call-ID. That means, after recvfrom there must be an extra processing finding the Call-ID header in message, to calculate a hash and then forward() message. The more the processing, the more cases when 200 could arrive before 180. I just changed it to round robin, and the amount decreased a lot, but it's still there. If I send a burst of 1000 messages, about 5 of them leave out of order every time. Best regards, Luis On 4/9/20 1:48 PM, Luis Rojas G. wrote: > Hello, > > I have a lot of experience developing mutithreaded applications, and I > don't see it so unlikely at all that a process loses cpu just after > recvfrom(). It's just as probable as to lose it just before, or when > writing on a cache or just before of after sendto(). If there are many > messages going through, some of them will fall in this scenario. if I > try sending a burst of 100 messages, I see two or three presenting the > scenario. > > Just forward() with a single process does not give the capacity. I'm > getting almost 1000caps. More than that and start getting errores, > retransmissions, etc. And this is just one way. I need to receive the > call to go back to the network (our application is a B2BUA), so I will > be down to 500caps, with a simple scenario, with no reliable > responses, reinvites, updates, etc. I will end up having as many > standalone kamailio processes as the current servers I do have now. > > I really think the simplest way would be to add a small delay to 200 > OK. Very small, like 10ms, should be enough. Simple and it should > work. As Alex Balashov commented he did for the case with ACK-Re-Invite. > > I have to figure out how to make async_ms_sleep() work in reply_route(). > > Thanks for all the comments and ideas > > Best regards, > > Luis > > > > . On 4/9/20 12:17 PM, Daniel-Constantin Mierla wrote: >> >> >> MICONDA(a)GMAIL.COM appears similar to someone who previously sent you >> email, but may not be that person. Learn why this could be a risk >> <http://aka.ms/LearnAboutSenderIdentification> >> Feedback <http://aka.ms/SafetyTipsFeedback> >> >> Hello, >> >> then the overtaking is in between reading from the socket and getting >> to parsing the call-id value -- the cpu is lost by first reader after >> recvfrom() and the second process get enough cpu time to go ahead >> further. I haven't encountered this case, but as I said previously, >> it is very unlikely, but still possible. I added the route_locks_size >> because in the past I had cases when processing of some messages took >> longer executing config (e.g., due to authentication, accounting, ..) >> and I needed to be sure they are processed in the order they enter >> config execution. >> >> Then the option is to see if a single process with stateless sending >> out (using forward()) gives the capacity, if you don't do any other >> complex processing. Or if you do more complex processing, use a >> dispatcher process with forwarding to local host or in a similar >> manner try to use mqueue+rtimer for dispatching using shared memory >> queues. >> >> Of course, it is open source and there is also the C coding way, to >> add a synchronizing mechanism to protect against parallel execution >> of the code from recvfrom() till call-id lock is acquired. >> >> Cheers, >> Daniel >> -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

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t_relay failure when dropping all branches in same destination set
by Sergiu Pojoga 09 Apr '20

09 Apr '20

Hi there, Can someone help figure out why there's this t_relay error, if I decide to drop the single master branch either all branches belonging to the same destination set, even though there are more serial sets to try later? *Error:* *tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -6 (-6)* *tm [t_funcs.c:355]: t_relay_to(): -6 error reply generation delayed * *Example:* request_route { seturi("sip:a@mydomain.net"); append_branch("sip:b@mydomain.net", "0.5"); append_branch("sip:c@mydomain.net", "0.5"); append_branch("sip:d@10.22.0.30", "1.0"); # append_branch("sip:e@mydomain.net", "1.0"); # no error if I add this branch with same Q=1.0 t_load_contacts(); t_next_contacts(); t_on_branch("check_branch"); t_on_failure("serial"); t_relay(); break; } branch_route[check_branch] { if($rd=="10.22.0.30") drop; } failure_route["serial"] { if (!t_next_contacts()) exit; } t_on_branch("check_branch"); t_on_failure("serial"); t_relay(); } Thanks in advance. --Sergiu

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