- sr-users - kamailio.org

Dispatcher latency stats monitoring with statsd
by Julien Chavanton 01 Dec '17

01 Dec '17

Not sure if anyone is using "latency measurement" feature in Kamailio dispatcher module, I thought I could share a very simple Python script that I am using to monitor remote gateway using this. Sample Script is attached, it will post data to statsd and is working in cron The feature can be enabled : 3.27. ds_ping_latency_stats (int) Enable latency measurement when pinging nodes If set to 0, disable latency measurement. If set to 1, enable latency measurement https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc… This is one use case of this feature, another one will be to load balance based on congestion detected on latency, I am planning to share a lab experiment in a little while. Regards

2 1

[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

Re: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue.
by otron2016＠gmail.com 29 Nov '17

29 Nov '17

Through Firefox likely means via websockets (wss) rather than via the sip port. Many things could interfere with the latter. Take heart, though: websockets is harder to set up so you're likely through the worst. Look at logs and see what iftop shows you about traffic on the server as you try to make sip connections. Also try your sip client with a known-good sip server (e.g. Iptel) just to make sure the problem isn't with the client. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/28/2017 6:39 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue. Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won’t work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don’t know 100% for sure. Thank you

1 0

Using SIP tools through Kamailio and Asterisk - chrome issue.
by Wilkins, Steve 28 Nov '17

28 Nov '17

Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won't work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don't know 100% for sure. Thank you

1 0

dmq replicate
by Aidar Kamalov 28 Nov '17

28 Nov '17

Hello, I set up kamailio cluster. dmq_usrloc works fine, but PUSBLISH and SUBSCRIBE replication not. If I make a call to first node I've got many(hundreds) PUBLISH messages from first node to first node and some messages (8 for example) to other nodes. And second xnotice in config not executes. And errors n log ERROR: <core> [core/sr_module.c:1726]: get_int_fparam(): Unsupported param type for int value: 32 How to replicate publish and subscribe, is it wrong way? my confiig: request_route { if(is_method("KDMQ")) { if ($Rp == 5050) { dmq_handle_message(); exit; } else { drop; } } ..... ..... if(is_method("PUBLISH|SUBSCRIBE")) { if(!dmq_is_from_node()) { xnotice("AAAAAAAA"); dmq_t_replicate("1"); xnotice("AAAAAAAABBBBBBBBBBB"); } } 2017/11/28 12:56:06.415601 192.168.8.213:5060 -> 192.168.8.213:5060 PUBLISH sip:34200@siptest. SIP/2.0 Via: SIP/2.0/UDP 192.168.8.213;branch=z9hG4bKc7a5.cee84051000000000000000000000000.0 To: <sip:34200@siptest> From: <sip:34200@siptest.>;tag=ae81df16b5a52fc1f62f9b45f337e660-db10 CSeq: 10 PUBLISH Call-ID: 4c8f9e6e799efcd7-57244(a)192.168.8.213 Content-Length: 556 User-Agent: kamailio (5.0.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Expires: 125 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:34200@siptest."> <dialog id="898932197-5060-86(a)BJC.BGI.D.DB" call-id="898932197-5060-86(a)BJC.BGI.D.DB" direction="initiator"> <state>Trying</state> <remote> <identity>sip:53003@siptest.t</identity> <target uri="sip:53003@siptest.tg"/> </remote> <local> <identity>sip:34200@siptest</identity> <target uri="sip:34200@siptest"/> </local> </dialog> </dialog-info> -- Aydar A. Kamalov

1 1

dispatcher/carrierroute and branches across SBC and remote peer
by Nathan Ward 28 Nov '17

28 Nov '17

Hi, We use Kamailio as a SIP server, and SEMS as an SBC. We have a need to use the dispatcher/carrierroute modules to provide HA/load sharing across two different parts of our environment: 1) Our SBCs 2) Our peer’s SIP servers We have pairs of SBCs (for now) and they have the same network connectivity - different IPs of course, but they can reach the same things. Our peer’s network has 2 SIP servers (again, for now) reachable through either SBC. We select which SBC pair to use based on where the calls are going, in a branch route. In some cases, we use parallel forking, in others we use serial forking (i.e. to a media server for a prompt, then sending the call to wherever it needs to go). Right now there is no automatic failover or load sharing, so this is all simple logic - if call is to x peer or y customer, use x SBC or y SBC. I’m trying to add the failover/loadbalancing now. I have a few half baked ideas about how to achieve a network like this, but thought I’d ask if anyone else had done this first before I went too overboard. Currently I’m thinking of calling cr_route to figure out which peer SIP server, from there figuring out which SBC pair, and then calling ds_select_dst to select which SBC in the pair to set $du. I’m being tripped up though, as I can’t call cr_route or ds_select_dst in a branch route, but, as far as I’m aware that’s where I’d need to do so in order to handle this for multiple branches. Should I be looping the request through the SIP server to achieve this (i.e. force $du = 127.0.0.1 in my branch_route), or, can I avoid that complexity? -- Nathan Ward

1 0

Kamailio didn't start before increasing fork_delay
by igor.potjevlesch＠gmail.com 27 Nov '17

27 Nov '17

Hello, We suddenly had an issue on one Kamailio instance: we were not able to restart. Kamailio started to boot and fork and suddenly crashed. The last logs reported a failure regarding the ability to connect to one MySQL instance. I finally succeed to restart after increasing: fork_delay=5000 to fork_delay=9000. How this could happen suddenly? We already restarted Kamailio on this server. Regards, Igor.

2 5

Does Kamailio support RFC4117?
by Ye, Charlie (NSB - CN/Qingdao) 27 Nov '17

27 Nov '17

Hi Friends, Does Kamailio support RFC4117? Do you have any suggestions if I want support RFC4117 through Kamailio? Thanks and Best Regards, Charlie

2 2

Dispatcher with kam & fs behind nat
by Daniel Greenwald 27 Nov '17

27 Nov '17

Dispatcher is sending inbound calls to Freeswitch boxes. Both the FS boxes and Kam are behind the same nat. What is the correct way to pass calls to the FS boxes so that all SIP between FS & Kam use local IP's? Should Kam put two record-route headers on, one with PUBLICIP and one with PRIVATEIP? How is this accomplished? Thanks

2 3

Kamailio 5.0 Android Calling Not working
by Zeeshan M 25 Nov '17

25 Nov '17

Hello Friends, I had successfully installed and configured Kamailio 5.0 on linux machine. And tried it with linphone app & following are the issues i am facing, 1) IOS to Android Calling not working with timeout error on server end when we checked logs 2) Android to Android calls also not working same error. Kindly let me know what could be the issue. Regards, Zeeshan.

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Re: [SR-Users] sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

I just verified that I do have logging in all pertinent areas. I added tls and I do see "TLS accept successful" in the kamailio logs, however, I see no log being generated in route[xhttp:request], and I do have logging in the first line of this route. I don't think it is getting there. Also, I noticed I have event_route[xhttp:request] not route[xhttp:request]. Thank you, -Steve

1 0

sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen's in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve

2 5

Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

Just a guess but maybe later entries [like +HIGH:+MEDIUM:+LOW] put it back. Try switching the order so that !ECDHE and the others you're trying to exclude come after. Sent from Samsung Mobile -------- Original message -------- From: Ilyas Keskin <ilyask92(a)gmx.de> Date: 11/24/2017 10:19 AM (GMT-08:00) To: miconda(a)gmail.com,"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite Hi Daniel, yes I am using the tls.cfg file. I tried your suggestion to add the cipher suite string (notice the !EDCHE which I also added to the httpd ssl.conf) but nothing changed. [server:default] method = TLSv1 cipher_list = !DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem certificate = /etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem #ca_list = ./modules/tls/cacert.pem #crl = ./modules/tls/crl.pem Also here is a log snippet from tls module section of kamailio initialization. Notice first two lines. Also it seems to me the module actually ignores the local openssl installation and uses its own which has been compiled with the module itself (?). Other than that it seems to be accepting the cipher_list value just fine: Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:355]: mod_init(): With ECDH-Support! Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:358]: mod_init(): With Diffie Hellman Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:587]: init_tls_h(): tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, compression: on Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:595]: init_tls_h(): tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, zlib compression: compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_ Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tls [tls_init.c:649]: init_tls_h(): tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 7864320 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 3932160 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_branch("MANAGE_BRANCH"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_reply("MANAGE_REPLY"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_failure("MANAGE_FAILURE"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSs<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSs<default>: certificate='/etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSs<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSs<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSs<default>: require_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSs<default>: cipher_list='!DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSs<default>: private_key='/etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSs<default>: verify_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSs<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:670]: set_verification(): TLSs<default>: No client certificate required and no checks performed Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSc<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSc<default>: certificate='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSc<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSc<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSc<default>: require_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSc<default>: cipher_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSc<default>: private_key='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSc<default>: verify_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSc<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:655]: set_verification(): TLSc<default>: Server MUST present valid certificate Would it be possible to compile the tls module with certain openssl config switches (i.e. no-ec no-dh)? Any other ideas? Best regards, Ilyas Keskin Am 24.11.2017 um 15:45 schrieb Daniel-Constantin Mierla: Hello, On 23.11.17 22:42, Ilyas Keskin wrote: Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. are you also using tls.cfg? If yes, there is an attribute for chiper list in it as well, try and see if works with it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Re: [SR-Users] sipML5 through kamailio
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

"So, tls is required for wss?" Yes, as of the last rfc. It can work without it if the client software (browser) will accept it though by now most will treat it as insecure. I forget whether sipML5 will but my guess would be no. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/24/2017 9:57 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hello SamyGo, So, tls is required for wss? I thought that when I set a listen to Kamailio-Server-IPAddress:10443 and I sent a request through that IP and Port, that Kamailio would at least see the request and attempt to do something with it. I will go add tls and try again. Thank you! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Can you clarify the IP addressing scheme as you've mentioned. There is no TLS interface? Advertised address for 5060 but not for WSS interface. Both listen and advertise are public IPs ? Also,you're handling WSS requests. Do you've xlog ines in the route[xhttp:request] to view when a request lands.! My config has this: #!substdef "!MY_IP_ADDR!123.134.156.167!g" #!substdef "!MY_WS_PORT!6010!g" #!substdef "!MY_WSS_PORT!6011!g" listen=tcp:MY_IP_ADDR:MY_WS_PORT listen=tls:MY_IP_ADDR:MY_WSS_PORT Then the xhttp event route: ``` event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); #Deny any HTTP requests on any port other than WS/WSS ports. if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } #Handle HTTP(s) onwards. xlog("L_INFO", "HTTP Request Received\n"); ``` On Thu, Nov 23, 2017 at 12:18 PM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hi Sammy, First of all, thank you for taking the time to respond. Yes, port 10443 is opened. I have used this port before as asterisk’s WebRTC port and iptables shows it as open. No, I can’t even get a registration using the configuration I listed. I have an xdbg log statement right after the request_route, and I see nothing. I do know that my xdbg logs are working though because, if I register or make a call using any sip tool, I see all my logging and everything works correctly. -Steve From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:00 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hi Steve, Can you confirm that port 10443 is reachable behind the NAT to Kamailio server, validate iptables too Does your SIPml5 demo client register successfully to Kamailio? are there enough xlog lines to print out if anything lands in Kamailio. Regards, Sammy On Thu, Nov 23, 2017 at 11:34 AM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen’s in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Cannot disable EC Diffie Hellman cipher suite
by Ilyas Keskin 24 Nov '17

24 Nov '17

Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. Best regards, Ilyas Keskin

2 2

Kamailio KEMI Framework Tutorial
by Daniel-Constantin Mierla 24 Nov '17

24 Nov '17

Hello, I started a tutorial for Kamailio KEMI Framework written in markdown format targeting use of mkdocs. It is published on github as part of kamailio-docs repository: * https://github.com/kamailio/kamailio-docs/tree/master/kamailio-kemi-framewo… The mkdocs build (to browse and read) is online at: * http://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/ It is work in progress and contributions from the community are more than welcome -- doing push requests on github project is the recommended way for contributions: * https://github.com/kamailio/kamailio-docs/ The main goal is document the role of KEMI, its benefits and how it can be used. It addition it should cover the list of functions exported to KEMI by core as well as special KEMI functions. Regarding the functions exported by kamailio modules to KEMI I added as an skeleton the ones exported by acc module. However, I am not sure if we should do it there or add them to the existing docs of the modules. Opinions on this?!? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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