sr-users

sr-users@lists.kamailio.org

42 participants
37517 discussions

Help on Video Conference
by Kranti Kumar 23 Oct '17

23 Oct '17

Hello , I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ My configuration details: HSS - 10.0.0.9 (./hss.sh) PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg) ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg) SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg) host machine: 10.0.0.5 I am able to register mobile clients ( IMSdroid) and make voice/video calls between them successfully. Now I would like to make a conference call among 3 IMSdroid users but its failing. Could you please suggest me the way forward here. I am thinking to integrate Application server (AS) to enable Video Conferences. Could you please guide me how to enable AS in this VM image and configuration details. Is there any possibility to do Video conference using this current setup(VM image). Could you please suggest me. Thank you so much in advance. Regards, -kranti

2 1

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

5 22

call_id in dialog module
by ‪mahsa _a_‬ ‪ 20 Oct '17

20 Oct '17

hi all;i want to write "billing" project with kamailio,i using python,postgresql, dialog module and falcon.my problem is in "call_id" section.my code(this section) is in python : """... if (method == 'INVITE'): callid = req.get_param_as_list("call_id") print(req.get_param_as_list("Ts")) print("call id = %s " % callid) elif (method == 'ACK'): time_req = int(req.get_param_as_list("Ts")[0]) callid = req.get_param_as_list("call_id")[0] print("user %s call to %s at time %d % (from_user,to_user,time_req))... """ and in kamailio.cfg: """http_query("http://127.0.0.1:5000/test?method=$rm& call_id=$dlg(callid)&Ts=$Ts&fu=$fu&tu=$tu","$var(result)"); """" "callid" in ACK is ok but "callid' in invite is "null", where is my problem? thanks.

2 1

Retransmission handling
by Volkan Oransoy 19 Oct '17

19 Oct '17

Hi all, I wonder purpose of this block in default script. What I didn’t understand is the purpose of "t_check_trans()”. Accoording to documentation, this method returns a boolean. So why do we use it even we don’t check output? # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } /Volkan

3 3

VoLTE Capable LTE Handsets which support IMS uisng IPv4
by Ron 19 Oct '17

19 Oct '17

I would like to support VoLTE using an IPv4 IMS network rather than IPv6 network. The handsets that I have been trying insist on the IMS network support either IPv6 or IPv4v6 dual stack. After giving them IPv4v6 connectivity, I am finding that even though the handsets asked for both IPv4 and IPv6 P-CSCFs, they only try the IPv6 proxy (if only the IPv4 proxy is provided, they will not try at all). I am wondering of anyone has worked with any LTE handsets which are willing to support IMS using IPv4 only. Thanks, Ron

3 2

Kamailio and FreeSwitch integration
by Timothy oladapo olawuyi 19 Oct '17

19 Oct '17

Dear All, I am a new entrant in Kamailio though little understanding of SIP and FreeSwitch soft switch, however, I have a project to integrate Kamailio and Freeswitch. Here is the scenario of what I want to implement; Kamailio will act as our SIP control while Freeswitch will act media server for incoming calls. Freeswitch will send all outgoing calls to Kamailio for onward transfer to our SIP provider network. No registration, presence, location Accounting, Authentication etc. are required. I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and Kamailio integration sample config available at http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still unable to figure out how to go with the configuration. I will appreciate any one who can provide guideline for achieving the above scenario. Regards Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims | <htttp://www.pethahiah.com/> htttp://www.pethahiah.com

3 4

port forwarding
by Andrzej Kaczmarczyk 19 Oct '17

19 Oct '17

Hello, This is probably trivial, but I'm quite new in kamailio and trying to handle it for my personal purposes (so - no trainings, no support etc :) ) - for few (up to 20) users and rare connections, more to learn something than to have production system. No plans to go outside my own server. I have LAN. One of machines in this LAN is server with kamailio onboard. It works, but only with local hostname (user@hostname). Now I want to go outside (with clients). As temporarty solution I have only dynamic DNS (no-ip.org) and I didn't expect it working. But tried with IP adress (user(a)WAN.IP.ADD.RESS). And... I don't know which ports to redirect on router. Now there are redirected 5060 and 10000-30000 (maybe it is not necessary but first I want it working) - taken from some on-line info. I can: - log in to all my accounts, - depends on client - call to another user and in some cases I can receive the call (so it is probably issue on my clients, because it is possible to make a call at all). But I cannot transmit any data: connections are completely silent in any case and I cannot start video - although cameras are findable. Which ports in fact are necessary? I want audio and video communication, presence info and probably nothing more. Which ports should I redirect? -- Pozdrawiam Andrzej Kaczmarczyk

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 19 Oct '17

19 Oct '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>

1 0

Problems with AAR over Rx Interface
by Ron McLeod 19 Oct '17

19 Oct '17

When a LTE handset client attempts to REGISTER with the IMS, an AAR is sent from the P-CSCF to the PCRF to establish a dedicated bearer for SIP signaling (as well as request notifications for certain access network event). There are a couple of problems with the flow descriptions, and I was hoping that someone could confirm that these are actual issues before I start working on a fix. 1. The "in" and "out" directions seem reversed. "in" corresponds to the uplink (UE to network), and "out" corresponds to the downlink (network to UE). In the AAR sent to the PCRF, the "in" description indicates the traffic "from" the UE, rather than "to" the UE. There is a similar problem with the "out" description. A fix would involve reversing "in" and "out". 2. The IP address and port number for the UE side are correct (in my case the UE has been allocated 172.25.152.195, and the client is using port 37843), but the network side is specified as IP address 127.0.0.1 and port number is the same as the open used by the client (in my case the P-CSCF has an IP address of 172.25.152.20 and a port number of 4060). I see in the source code that the P-CSCF address is hard-coded to "127.0.0.1". A fix would involve either using the actual UE-facing IP address and port number for the P-CSCF or wildcards. Another option would be to not try and setup a dedicated bearer for SIP signaling, and just use the default bearer (there shouldn't be anything else other than signaling-related traffic on the default bearer anyway). The access network HSS would specify that the default bearer was created with the appropriate parameters (QCI=5, AM, etc.). This is the network message flow: 172.25.152.195 -> pcscf.core.ims1.test SIP 999 Request: REGISTER sip:provider1.test (fetch bindings) | pcscf.core.ims1.test -> 172.25.152.195 SIP 359 Status: 100 Trying (0 bindings) | pcscf.core.ims1.test -> pcrf.ims1.test DIAMETER 762 cmd=AARequest(265) flags=RP-- appl=3GPP Rx(16777236) h2h=28fe20b1 e2e=fe338ed9 pcrf.ims1.test -> pcscf.core.ims1.test DIAMETER 198 cmd=AAAnswer(265) flags=-P-- appl=3GPP Rx(16777236) h2h=2 e2e=13 pcscf.core.ims1.test -> 172.25.152.195 SIP 410 Status: 408 Request Timeout (0 bindings) | This is the DIAMETER AAR message: Diameter Protocol Version: 0x01 Length: 696 Flags: 0xc0, Request, Proxyable Command Code: 265 AA ApplicationId: 3GPP Rx (16777236) Hop-by-Hop Identifier: 0x28fe20b1 End-to-End Identifier: 0xfe338ed9 AVP: Session-Id(263) l=40 f=-M- val=pcscf.core.ims1.test;548405219;1 AVP: Origin-Host(264) l=28 f=-M- val=pcscf.core.ims1.test AVP: Origin-Realm(296) l=22 f=-M- val=provider1.test AVP: Auth-Application-Id(258) l=12 f=-M- val=3GPP Rx (16777236) AVP: Vendor-Specific-Application-Id(260) l=32 f=-M- AVP: Destination-Realm(283) l=22 f=-M- val=core.ims1.test AVP: Subscription-Id(443) l=52 f=-M- AVP: Media-Component-Description(517) l=304 f=VM- vnd=TGPP AVP Code: 517 Media-Component-Description AVP Flags: 0xc0 AVP Length: 304 AVP Vendor Id: 3GPP (10415) Media-Component-Description: 00000206c0000010000028af0000000100000207c00000bc... AVP: Media-Component-Number(518) l=16 f=VM- vnd=TGPP val=1 AVP: Media-Sub-Component(519) l=188 f=VM- vnd=TGPP AVP Code: 519 Media-Sub-Component AVP Flags: 0xc0 AVP Length: 188 AVP Vendor Id: 3GPP (10415) Media-Sub-Component: 000001fdc0000010000028af00000001000001fbc0000046... AVP: Flow-Number(509) l=16 f=VM- vnd=TGPP val=1 AVP: Flow-Description(507) l=70 f=VM- vnd=TGPP val=permit out ip from 172.25.152.195 37843 to 127.0.0.1 37843 AVP: Flow-Description(507) l=69 f=VM- vnd=TGPP val=permit in ip from 127.0.0.1 37843 to 172.25.152.195 37843 AVP: Flow-Usage(512) l=16 f=VM- vnd=TGPP val=AF_SIGNALLING (2) AVP: Media-Type(520) l=16 f=VM- vnd=TGPP val=CONTROL (4) AVP: Codec-Data(524) l=28 f=VM- vnd=TGPP val=downlink\noffer\n AVP: Codec-Data(524) l=27 f=VM- vnd=TGPP val=uplink\nanswer\n AVP: Flow-Status(511) l=16 f=VM- vnd=TGPP val=ENABLED (2) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=CHARGING_CORRELATION_EXCHANGE (1) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_LOSS_OF_BEARER (2) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_RECOVERY_OF_BEARER (3) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_RELEASE_OF_BEARER (4) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_ESTABLISHMENT_OF_BEARER (now void) (5) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=IP-CAN_CHANGE (6) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=ACCESS_NETWORK_INFO_REPORT (12) AVP: Framed-IP-Address(8) l=12 f=-M- val=172.25.152.195 AVP: Authorization-Lifetime(291) l=12 f=-M- val=7200 AVP: Auth-Grace-Period(276) l=12 f=-M- val=0 AVP: Session-Timeout(27) l=12 f=-M- val=7200

2 1

Problem removing headers from FAILURE_ROUTE
by Joel Serrano 17 Oct '17

17 Oct '17

Hi all, I have the following scenario: Client -> Kamailio (adds X-CallSource hdr) -> Media Server -> Kamailio (removes all "X-" headers) -> Carrier Carrier doesn't get any X- headers. All works. Now for example, if the media server replies with a non-OK status, like: Client -> Kamailio (adds X-CallSource hdr) <-> Media Server (480 Temporarily Unavailable) Kamailio (in failure_route, removes all "X-" headers) -> External Voicemail Service External Voicemail Service gets the X- headers. Basically, on the first example, it is a regular call, all handled from request route. On the second example, call goes to media server, media server replies with a 480, reaches kamailio's failure route, in there we run the same remove_hf_re. In the captures I can see that the X-CallSource header (added at the very beginning) is still there. I have tried different combinations in the failure route: a) Tried removing exactly the header: remove_hf("X-CallSource") b) Tried removing any X- header: remove_hf_re("X-") Those didn't work, then I tried: c) append_hf("X-CallSource: Test") remove_hf("X-CallSource") The result was now two X-CallSource headers, the add worked, the remove didn't. I thought maybe "msg_apply_changes()" could help, but it isn't valid inside failure_route. So what is the deal here? Am I doing something completely wrong or is it just that you cannot remove headers (but you can add) in failure route? According to the doc: https://www.kamailio.org/docs/modules/devel/modules/textops.html#textops.f.… -----> This function can be used from ANY_ROUTE. OS: Debian 8 Kamailio: 4.4.5 (installed from deb repo) # kamailio -v version: kamailio 4.4.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 Any suggestions? Thanks Joel.

2 4

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users