sr-users November 2019

sr-users@lists.kamailio.org

67 participants
100 discussions

by Kjeld Flarup

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN IP4 192.168.237.239 Media Attribute (a): rtpmap:96 H264/90000 Media Attribute (a): fmtp:96 profile-level-id=42000d;packetization-mode=1 Media Attribute (a): sendrecv After sdp_remove_codecs_by_id I send this. But without "96" Media Description, name and address (m): video 6534 RTP/AVP Media Attribute (a): sendrecv Media Attribute (a): rtcp:6535 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): candidate:r9KO0wNPMTGktklS 1 UDP 2130706431 192.168.237.15 6534 typ host Media Attribute (a): candidate:r9KO0wNPMTGktklS 2 UDP 2130706430 192.168.237.15 6535 typ host My problem is that the door telephone does not do an ACK when it gets the 200 OK My suspicion is that it does not understand the video response Media Description, name and address (m): video 0 RTP/AVP 0 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): sendrecv Is there a way to remove the whole video part and not just the coded id? It is not an option to disable video in the door telephone, as we need the video. It is just in the fallback is must be removed. -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 years, 12 months

Kamailio does not detect message

by Adesh Pandey

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Kamailio version: 5.2

4 years, 12 months

Kamailio developers meeting - follow up remarks

by Daniel-Constantin Mierla

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 years, 12 months

Kamailio xavp_contact - Segmentation Fault

by Ilie Soltanici

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you.

4 years, 12 months

Kamailio Parrallel Forking

by Ilie Soltanici

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "Contacts loaded for $rU\n"); } if (!t_next_contacts()) { xlogl("L_INFO", "t_next_contacts - Only one contact found for $rU, calling\n"); } else { xlogl("L_INFO", "t_next_contacts - Multiple contacts found, parallel forking\n"); } route(RELAY); ------------------------------------------------------------------------------------------------------------------------------------ I noticed that on Kamailio 5.2.5 - this code is working fine, and Kamailio is sending an INVITE to both Devices.But, on Kamailio 5.3.1 - that's not working anymore. For some reason, kamailio is sending just one INVITE to one device with wrong URI. The URI is set to the second device instead of the correct one. Was something changed in the t_next_contacts function related to this? Thank you

4 years, 12 months

Kamailio 5.2.3 segfaulting

by Daniel Greenwald

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

4 years, 12 months

No Audio using RTP Proxy

by Sujit Roy

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with 192.168.100.177:27360 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with 192.168.100.177:27361 But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179 What can be done to change the caller's RTP address to x.179 in RTPProxy ? Thanks -- Regards =================== Sujit Roy

4 years, 12 months

Encode host ip in Contact header for semi-stateless setup

by Daniel Greenwald

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the same kamailio by not altering the VIA but allow the subsequent in-dialog transactions to potentially hit another kamailio by changing the contact to a shared SRV record (and encoding the freeswitch box's IP into it for later retrieval.) Thanks in advance. Daniel Greenwald

4 years, 12 months

dialog module: Does dlg_manage() have to be called for all requests?

by George Diamantopoulos

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); ... } In this example code, dlg_manage seems to be called for all requests, as it is not dependent on the conditional that only requests with no to-tag are to be handled. Previously, Daniel had suggested to me in person that one also run dlg_manage() for CANCELs, and in-dialog reINVITEs, BYEs and ACKs to mitigate some buggy corner cases. Is this still the case? What do you guys and gals do in production? Have you had to revise how you use dlg_manage in terms of calling it for initial INVITEs only vs all requests? Thanks! BR, George

4 years, 12 months

Modify SIP Message Header and Message Body parameter

by Sujit Roy

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.1681.101:5060;rport=5060;branch=z9hG4bK5a61798d7c850d39 Record-Route: <sip: 192.1681.102;lr> From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567 @ 192.1681.102>;tag=as728725b3 Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE Server: VOS3000 V2.1.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 600;refresher=uas *Contact: <sip:1018801751481417@104.216.39.66:5060 <http://sip:1018801751481417@104.216.39.66:5060>>* Content-Type: application/sdp Require: timer Content-Length: 289 v=0 o=root 912643161 912643161 IN IP4 192.1681.104 s=VOS3000 V2.1.4.0 c=IN IP4 192.1681.104 t=0 0 m=audio 10784 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv *a=nortpproxy:yes* 2. I Want to hide the Contact Parameter and a= nortpproxy:yes ==> How to do it as well ? Thanks in advance. -- Regards =================== Sujit Roy

4 years, 12 months

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sr-users November 2019