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[Kamailio]list members
by Djamel Bahamid 01 Aug '14

01 Aug '14

Hi, I would like to subscribe to the list. Best Regards, Djamel BAHAMID.

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SIGUSR1 for memory status not working as documented - only one process reports back
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a kamailio process will dump a report of the allocation map when shutting down, or when receiving a SIGUSR1. I have configured my kamailio.cfg with memlog=1 and no other change, and I see the memory report on shutdown for all processes. However, when I send a SIGUSR1 to a kamailio process, the process does absolutely nothing, with the exception of the first kamailio process (the one reported as Type=attendant by "kamctl ps"). All of the other processes just ignore SIGUSR1. What is going on? It is inconvenient to force a shutdown of all the kamailio processes just to get the memory report.

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Unable to SUBSCRIBE for presence using sip.js through WSS
by Alex Villacís Lasso 31 Jul '14

31 Jul '14

My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the NOTIFY for the presence event. However, MESSAGE packets are routed correctly between peers, even through websockets. The location table after registration looks like this: +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | id | ruid | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods | instance | reg_id | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | 192 | uloc-53d96398-1afc-1 | admin | pbx.villacis.com | sip:nfinhooe@192.0.2.90;transport=ws | sip:192.168.3.2:41478;transport=WS | NULL | 2014-07-30 17:18:57 | -1.00 | 6a2dot9ihoobhng9ackqmq | 90 | 2014-07-30 17:08:57 | 0 | 64 | SIP.js/0.5.0 | tls:192.168.2.18:5061 | 782 | <urn:uuid:b8888e95-7833-46a4-a21e-8d2e809a66b8> | 1 | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ When running the following javascript code: var subs = ua.subscribe('admin(a)pbx.villacis.com', 'presence'); subs.on('notify', function (n) { alert(n.request.body); }); ... I see the following traffic in the browser console area: SUBSCRIBE sip:admin@pbx.villacis.com SIP/2.0 Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279 Max-Forwards: 70 To: <sip:admin@pbx.villacis.com> From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="admin", realm="pbx.villacis.com", nonce="U9lswVPZa5WSu5jz1P/jIOJTvthKXw6v", uri="sip:admin@pbx.villacis.com", response="f7246e64573b5888d69896a70d4204aa" Event: presence Expires: 3600 Contact: <sip:nfinhooe@192.0.2.90;transport=ws> Allow: ACK,CANCEL,BYE,OPTIONS,MESSAGE Supported: outbound User-Agent: SIP.js/0.5.0 Content-Length: 0 SIP/2.0 202 OK Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279;rport=41478;received=192.168.3.2 To: <sip:admin@pbx.villacis.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-84c3 From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.2.18:5061;transport=ws> Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 However, I also get the following messages in the kamailio log: Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription I have found a discussion at http://marc.info/?l=sr-users&m=136624316816288&w=2 which recommends fix_nated_contact(), but I am not really sure about how to add it, or what conditions to check on whether it should be used. I added the call to fix_nated_contact() like this: # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); # # 1 - Contact header field is searched for occurrence of RFC1918 or rfc6598 addresses. # 2 - the "received" test is used: address in Via is compared against source IP address of signaling # 16 - test if the source port is different from the port in Via # 64 - test if the source connection of signaling is a WebSocket # ----- # 83 #!ifdef WITH_WEBSOCKET # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. if (nat_uac_test("83")) { #!else if (nat_uac_test("19")) { #!endif if (is_method("REGISTER")) { fix_nated_register(); #!ifdef WITH_WEBSOCKET if (nat_uac_test("64")) fix_nated_contact(); #!endif } else { if (is_first_hop()) if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); } #!endif return; } After registration, "kamctl ul show" reports this: [root@elx3 js]# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: admin(a)pbx.villacis.com Contact:: sip:85017jar@192.168.3.2:41777;transport=ws Q= Expires:: 584 Callid:: argl25kg0r4qe8lbtbhn9f Cseq:: 82 User-agent:: SIP.js/0.5.0 Received:: sip:192.168.3.2:41777;transport=WS State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: tls:192.168.2.18:5061 Methods:: 782 Ruid:: uloc-53d971c7-297f-1 Instance:: <urn:uuid:75464840-96b8-4e22-8eaa-5012ba327086> Reg-Id:: 1 Last-Keepalive:: 1406759390 Last-Modified:: 1406759390 However, it still does not work: Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription

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Obscuring SIP traffic and using with NoSIP
by Muhammad Shahzad 31 Jul '14

31 Jul '14

Hi, As the mobile voip is getting more and more popular these days, there has been a strong opposition from GSM operators against mobile voip apps. They often use tactics like blocking voip ports, or detect and block voip traffic and in some cases restricting udp traffic altogether to very low upload and download speeds. See below link for some details, http://www.linphone.org/eng/blog/linphone-over-3g.html While not all the problems can be solved right now (especially the limiting udp traffic, since RTP always uses udp transport) I was wondering if we can at least handle the sip related problems. The most important of them is SIP traffic detection. While some forks would suggest using TCP/TLS to encrypt SIP traffic, it has a few problems, e.g. 1. It requires somewhat high resources on mobile devices, so many low-end android phones simply can't use it. 2. There is possibility that encryption signature may identify it as SIP traffic. There exists firewalls (often deployed in middle eastern countries) which have huge database of encryption signatures and patterns which although may not decrypt the sip packet but at least identify it as sip packet and block it. Also with rough agencies of evil empires spying over millions of users worldwide makes the current encryption standards pretty much pointless, at least in terms of user privacy and network security. So there is a strong need to experiment with new ideas and concepts to regain internet freedom. Some of such ideas are, 1. Convert sip traffic which is plain text to binary format just before transmitting it and revert it to plain text upon reception. 2. XOR the sip traffic (pretty much same as binary sip). 3. Use some very lightweight but effective / non-standard encryption algorithm, e.g. https://github.com/mshary/itv All these ideas require that SIP server such as Kamailio is able to adopt to these, preferably with minimal or no change in native code. The NoSIP module seems an interesting module in this regard. It provides all traffic it thinks is not the SIP traffic to configuration script, where we can do our own parsing and do whatever we want with it. I have two questions about this, 1. If parsed message is SIP, we can we send it back to kamailio core to get it processed as if it is a normal SIP message received by kamailio? 2. Can this module or any other module available in kamailio, that can provide us full sip packet that is about to be transmitted over sip socket, so we can "encode" it just before it is sent to next hop? I know this would be like writing a SIP transport in kamailio script which would be very tough if not impossible to implement in native core. But it will really help in winning the modern mobile voip challenges. Thank you.

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[Serusers] how to enable logs in SER
by pawan 31 Jul '14

31 Jul '14

Sir pl help for registration of soft phone to my local sip server my email vijay8538(a)gmail.com --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

1 0

multi-homed rtpengine
by Paul Belanger 30 Jul '14

30 Jul '14

Greetings, We have had good success in the past using rtpproxy with internal and external interfaces however, we are attempting to migrate to rtpengine now but are running into an issue (apparently this feature is not supported). So, my question is as follows, does this mean we need to multiple rtpengine and bind them to each interface? Additionally, in this setup can audio be bridged between the 2 rtpengine processes? If not, how do people over come this difference? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

1 0

Re: [SR-Users] LCR
by Keith 30 Jul '14

30 Jul '14

Hi, I want to do LCR based on from uri. I get I need to do regular expression matching but have no idea where to look to start. Any pointers would be good. Thanks Keith

3 2

Warning message on SIP websocket
by Alex Villacís Lasso 29 Jul '14

29 Jul '14

Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Via: SIP/2.0/TLS 192.168.3.2:58657#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: Er/CgVyBNzVGREBmt+w3FTff8s4=#015#012Server: kamailio (4.1.4 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012>

2 1

Plz help: MSILO on TLS + cert based client auth enabled server
by Marc M. 29 Jul '14

29 Jul '14

Hi, I have a SIP server with TLS + client certificate based authentication. The server is listening for SIP(UDP) on port 5060, and for SIPS(TLS) on port 5061) The client certificate based authentication is enforced on port 5061. I would like to fire up MSILO and face now a strange problem: At each REGISTER, the stored messages get dumped. Dumped messages get sent to the clients from senders logged in over port 5060(UDP) as SIP:xxxxxxxxx@xx.xx.xx.xx however Dumped messages are NOT sent to clients from senders logged in on port 5061(TLS) as SIPS:xxxxxxxx@xx.xx.xx.xx It looks like the messages get dumped so M-dump() is working, but they will never enter the route. I would assume TLS is stopping the dumped messages to entering the route logic. I would either 1. somehow force m_dump() to pipe the messages over SIP(and not SIPS), or 2. disable client auth for messages recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc

2 1

Only a redirect server
by Jack Smith 29 Jul '14

29 Jul '14

Dear friends, Just new to this mailing list, new to Kamailio, and new to running a sip server. I would like to try only a redirect server. I had a look at http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg which was friendly, but then the default file kamailio.cfg looked frightening ! Could someone help me set up a minimal Kamailio installation only to handle state-less redirect? Best, Jack

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