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[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 30 Nov '17

30 Nov '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM18/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2018/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 3rd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

3 2

blf in kamailio dmq cluster
by Aidar Kamalov 30 Nov '17

30 Nov '17

Hello, I have two(or more) kamailio servers with dmq module and dmq_usrloc - sip1.example.com and sip2.example.com publish and subcribe replication works too. BLF works fine if clients connected to the same kamailio server, because in active_watchers table stored with same domain sip1.example.com But what to do when call going from one server to second? Is there any configuration to ignore domain in pusblish, subscribe, notify, etc? -- Aydar A. Kamalov

3 7

OpenSIPS-devel not starting (TCP Error)
by SamyGo 29 Nov '17

29 Nov '17

Hi, I just installed the head version from git. version: opensips 2.4.0-dev (x86_64/linux) flags: STATS: On, SHM_EXTRA_STATS, EXTRA_DEBUG, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: d1a4419 main.c compiled on 19:53:20 Nov 28 2017 with gcc 4.8 It was compiled with EXTRA_DEBUG and DBG_TCPCON flags; While starting opensips with the residential vanilla script it gives following error: Nov 28 20:05:52 [30899] ERROR:core:tcp_init: oom con hist Nov 28 20:05:52 [30899] CRITICAL:core:main: could not initialize tcp Nov 28 20:05:52 [30899] INFO:core:cleanup: cleanup Nov 28 20:05:52 [30899] DBG:core:shm_mem_destroy: destroying the shared memory lock Nov 28 20:05:52 [30899] NOTICE:core:main: Exiting.... already running Even if I remove the proto_tcp, and tcp related things this still gives error and doesn't start. Kindly help. Regards, Sammy

2 3

parse_from_header(): bad From header
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. We experience the problem with malformed messages. First of all, I want to say, that most of time kamailio works well and nothing happens that can drop sessions. But from time-to-time something changes (may be in the network) and kamailio receives requests with malformed headers (To or From hfs). The schema: uplinks -> kamailio <-> routing server (asterisk) Malformed messages were received only from asterisk server. Sip debug on asterisk showed that messages were transmitted correctly and headers were nice. But kamailio obtains changed (broken) packets or perhaps can't read them properly. How does it look like: Nov 10 12:37:06 kamailio-name kamailio[965]: INFO: <script>: Going to NATMANAGE for BYE from sip:useragent7122@kamailio_address:5068 (IP=wss:client_address:62881) - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: NOTICE: <script>: Relaying request to <null> - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: uac [replace.c:783]: restore_uris_reply(): failed to find/parse FROM hdr Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: Skip manage DEVICE_STATE for BYE from <null> (IP=udp:asterisk_address:50600) with (200 - OK) - R=<null> ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: ------------------------------------MANAGE BYE by DEVICE_STATE_BYE_MANAGE route - 200 - OK M=BYE IP=udp:asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine_funcs.c:331]: get_from_tag(): failed to parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine.c:2252]: rtpp_function_call(): can't get From tag Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header How the packet looks like: ACK sip:tt7etmau@bb0pd5t63b26.invalid;alias=client_address~57573~6;transport=ws SIP/2.0 Via: SIP/2.0/UDP asterisk_address:50600;branch=z9hG4bK117106ca;rport Route: <sip:kam2.domain.com:5068 ;nat=yes;transport=udp;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331>,<sip:kam2.callision.com:5068 ;nat=yes;transport=ws;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331> Max-Forwards: 70 From: "19172423539" <sip:19172423539@asterisk_address:50600>;tag=as06869a5d To: <sip:useragent7122@10.0.1.18:5068>;tag=nmv7kmpau3 Contact: <sip:19172423539@asterisk_address:50600> Call-ID: 629d5a132c194c536b5f5c1a2a3c6e32@asterisk_address:50600 CSeq: 102 ACK User-Agent: asterisk PBX Content-Length: 0 The transport is used to send messages between asterisk and kamailio is TCP. TCP configurations: tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 I found the idea, that we need to change following parameters: modparam("uac","restore_mode","auto") modparam("uac","restore_dlg",1) to: modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) I did that, now it looks like all is fine, but I think it can get back. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua>

3 5

Source socket/port for in-dialog requests
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. My apologies for so frequent appealing to you. I'm trying to solve problem with ending of sessions. The problem consists of no 200 OK coming from uplink on our BYE requests. Topology. First leg: Webrtc client <-wss-> kamailio <-sip tcp-> asterisk routing server Second leg: Uplink <-sip tcp-> kamailio <-sip tcp-> asterisk routing server The problem appears only in case when dialog was ended by webrtc client. The first leg (dialog) of the call ends nice, without any hints on problem. But the second leg (with uplink) has problem with no 200 OK (coming from uplink) on BYE request coming from asterisk. Handshake between asterisk server and uplink establishes properly. It looks like: uplink.domain.com:5060 -> INVITE tcp -> our.kamailio.server:5060 -> INVITE tcp -> our.asterisk.server:5060 . uplink.domain.com:5060 <- 100 trying tcp <- our.kamailio.server:5060 . uplink.domain.com:5060 <- 180 ringing tcp <- our.kamailio.server:5060 <- 180 ringing tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 <- 200 OK tcp <- our.kamailio.server:5060 <- 200 OK tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 -> ACK tcp -> our.kamailio.server:5060 -> ACK tcp -> our.asterisk.server:5060 . uplink.domain.com <-media stream-> rtpengine <-media stream-> our.asterisk.server . Here kamailio starts to use random source port for relaying in-dialog BYE uplink.domain.com:5060 <- BYE tcp <- our.kamailio.server:45355 <- BYE tcp <- our.asterisk.server:5060 . And here the leg with uplink is expected to end with 200 OK (coming from uplink proxy). But uplink doesn't answer at all. We requested a tcpdump from uplink to see how packets are forwared from their side. And I saw that that 200 OK tries to be sent within first tcp session - to 5060 port of kamailio server. But sngrep on our side shows nothing, nothing appears in kamailio log, so 200 OK can't reach the 5060 socket because of some transport problem I think. Top via hf of our BYE request, contains kamailio record = uplink.domain.com:5060 My main thought on this count is to suppress using of random source ports for in-dialog requests, this behaviour looks irrelevant. We have turned on mhomed parameter (mhomed=1). And cookbook says: "Set the server to try to locate outbound interface on multihomed host. This parameter affects the selection of the outgoing socket for forwarding requests." But, there is a big but - we can not turn of this parameter for this moment, because routing script made with using of this one. Kamilio works on AWS EC2 platform. It has following IP address schema: private address atouched to container -> public address assinged to this private address (so this is standard ip address schema for AWS containers). For this example private address will be: 10.0.0.1 and public address will be: 100.0.0.1 Configuration - main parameters (all values are changed for an example): advertised_address=our.kamailio.server advertised_port=5060 alias=our.kamailio.server alias=our.kamailio.server:5060 alias=100.0.0.1 alias=100.0.0.1:5060 alias=10.0.0.1 alias=10.0.0.1:5060 auto_aliases=no port=5060 listen=100.0.0.1 listen=10.0.0.1 listen=tcp:100.0.0.1:5060 listen=tcp:10.0.0.1:5060 mhomed=1 fork=yes fork_delay=5000 children=6 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 One of the solutions to this problem was to add following inside the relaying route block: $fs = "tcp:" + "10.0.0.1" + ":5060"; if (!t_relay()) { sl_reply_error(); } But it seems to me it's not a smart solution. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 1

Best way to do routing against a list of gateways
by voipspace voipspace 29 Nov '17

29 Nov '17

3 6

allow_trusted not passing
by voipspace voipspace 29 Nov '17

29 Nov '17

4 5

URL problems connecting to external (Azure) MySQL database
by Floimair Florian 29 Nov '17

29 Nov '17

Hi! I'm having troubles connecting Kamailio to an external MySQL database (on Microsoft Azure Cloud). Initially creating the database with kamctl worked fine, but accessing it at runtime via db_url fails. A somewhat unusual thing about the connection to a MySQL database on Azure is that the username contains an '@' character. This leads me to believe that this case is somewhat unhandled when parsing the database URI. Trying to connect to something like modparam("sipcapture", "db_url", "mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db") Results in errors starting Kamailio ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'clouddb' (-2) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: sipcapture [sipcapture.c:1146]: child_init(): unable to connect to database [mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db] from capture_mode param. Is there a trick on how to achieve a connection in this case like some sort of escape character probably? Mit freundlichen Grüßen Florian Floimair (IMS) 1568

2 4

lcr_id vs lcr_count
by Patrick Wakano 29 Nov '17

29 Nov '17

Hello Kamailio list! I've been working with the LCR module and I wanted to use the lcr_id to match some ID's I have somewhere else. However, due to the lcr_count my lcr_ids must be less than the lcr_count. I understand the reason behind the lcr_count, what I think is a mistake is to have the lcr_id dependent of the lcr_count. Ideally we should be able to use any lcr_id value, just observing that the total amount of lcr groups should not exceed the lcr_count. The problem is this if that appears in the load_gws, to_gw and from_gw functions in lcr_mod.c: if ((lcr_id < 1) || (lcr_id > lcr_count_param)) { LM_ERR("invalid lcr_id parameter value %d\n", lcr_id); return -1; } I ended up using another table to make the conversion of ids, but this is horrible.... Any other reason to force the lcr_id to be less than the lcr_count? Best Regards, Patrick Wakano

3 3

how to add new media server in dispatcher table through "ANSIBLE"
by Narayan P 29 Nov '17

29 Nov '17

Hello Experts, I am new to ansible and deploying kamailio through ansible.Sucessfully deployed kamailio but my requirement is to add media server through ansible.As I am using dispatcher module in kamailio,not able to find any script that directly insert one media server IP and port into dispatcher table from ansible.(I am using ansible in one system and deploying kamailio in other). Any suggestion will be highly appreciated as always. Thanks, Narayan +91-9035438928

3 4

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