sr-users October 2012

sr-users@lists.kamailio.org

87 participants
164 discussions

Unknown status in presence_dialog
by David 29 Oct '12

29 Oct '12

Hello, I am using the presence_dialoginfo module on a stand alone presence server. I have this : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; If the server does not have any status information, it will send out a blank notify with a 0 content length. Is there anyway to tell Kamailio to send an xml doc with status "terminated" if the server does not know the status that was requested by the SUBSCRIBE? Thanks, David Server : version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

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siptrace vs. sipcapture
by Mino Haluz 29 Oct '12

29 Oct '12

Hi, we are thinking about switching from siptrace to sipcapture + HEP encapsulation. Do you know some reasons why we should switch to sipcapture or why we should not ? ;) We have multiple SIP proxies and Cisco gateways. In earlier versions of kamailio was siptrace blocking, that's why we started to looking for some other capturing mechanism. I think now siptrace is buffered, so we are hesitating .. Mino

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rtpproxy ipv4/ipv6 bridging question
by Juha Heinanen 29 Oct '12

29 Oct '12

is it possible to make rtpproxy to include both ipv4 and ipv6 address in sdp? if not, i don't see how ipv4/ipv6 bridging can be made to work when $rd is domain name. -- juha

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NEW: new module: xhttp_pi
by Ovidiu Sas 28 Oct '12

28 Oct '12

Hello all, A new module is available in trunk. It provides a web provisioning interface: http://kamailio.org/docs/modules/devel/modules/xhttp_pi.html A sample config example is provided in the source tree: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… An additional xml file is required for web config framework (what to configure, where and how). A sample xml file is provided in the source tree for dispatcher and dialplan tables: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… When kamailio is up and running, a web provisioning interface for managing db records will be available. No supplementary web server needs to be installed in order to manage the db records. Also, it provides a tight control over what can be modified and how. Testing and feedback is appreciated. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

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web homer not displaying data
by Christopher Holland 26 Oct '12

26 Oct '12

I have sipcapture and siptrace setup on the same server using kamailio 3.4.0-dev4 and webhomer 3.3.0 setup to capture sip traffic from a mirriored port on the LAN. Interface ETH1 is to capture and ETH0 is for managment. The sipcapture database is getting data put into it. I can log into webhomer, but see no data in the graphes nor can I search. I have double checked my database permissions and they seem to be set up correctly. webserver is set up with: Apache/2.2.16 (Debian) PHP5 MYSQL Json phpmyadmin Here is a copy of my kamailio.cfg and configuration.php rev_dns=no auto_aliases=no debug=1 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=5 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* IP and port for HEP capturing) */ listen=udp:127.0.0.1:9060 mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "siptrace.so" # ----- mi_fifo params ----- ####### Routing Logic ######## modparam("sipcapture", "db_url", "mysql://openser:******@localhost/homerdb") modparam("sipcapture", "capture_on", 1) /* activate HEP capturing */ modparam("sipcapture", "hep_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.0.0:5060-5080") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth1") /* activate IPIP capturing */ #modparam("sipcapture", "raw_ipip_capture_on", 1) /* My table name*/ modparam("sipcapture", "table_name", "sip_capture") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* activate monitoring/mirroring port capturing. Linux only */ modparam("sipcapture", "raw_moni_capture_on", 1) /* Promiscious mode RAW socket. Mirroring port. Linux only */ modparam("sipcapture", "promiscious_on", 1) /* activate Linux Socket Filter (LSF/BPF) on mirroring interface. Linux only */ #modparam("sipcapture", "raw_moni_bpf_on", 1) /* sipcapture node */ modparam("sipcapture", "capture_node", "homer01") # check IP and port of your capture node modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag",22) modparam("siptrace", "trace_on", 1) # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { sip_capture(); drop; } onreply_route { sip_capture(); drop; } ################################## configuration.php <?php /*********************************************************************************/ /* Access db of homer */ define('HOST', "localhost"); define('PORT', 3306); define('USER', "webhomer"); define('PW', "******"); define('DB', "web_homer"); /* Homer connection * this user must have the same password for all Homer nodes * please define all your nodes in homer_nodes table */ define('HOMER_HOST', "localhost"); /* DEFAULT. Don't forget insert this host to your DB nodes table */ define('HOMER_PORT', 3306); define('HOMER_USER', "openser"); define('HOMER_PW', "******"); define('HOMER_DB', "homerdb"); define('HOMER_TABLE', "sip_capture"); /*********************************************************************************/ /* webHomer Settings * Adjust to reflect your system preferences */ define('PCAPDIR',"/var/www/webhomer/tmp/"); define('WEBPCAPLOC',"/webhomer/tmp/"); define('APIURL',"http://localhost"); define('APILOC',"/webhomer/api/"); /* INCLUDE preferences */ include("preferences.php"); ?>

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Message Routing To/From Gateway
by Ed Brady 26 Oct '12

26 Oct '12

Hi, I am trying to route SIP messages to a gateway. I am using SIP MESSAGE over UDP as well as the MSILO module for storing messages for users that aren't currently logged in. I have read through the Dynamic Routing module info but am not able to determine the best way to accomplish the routing I require. Our scenario is outlined below. User A sends a message addressed to "sip:12345678900@mydomain.com" Kamailio at mydomain.com checks if 12345678900 is a registered user in the system. If not, the message header is translated to "sip:12345678900@gatewaydomain..com" and forwarded. Incoming messages need to be accepted from their IP as well. It would also be useful if the address came as "sip:2345678900@mydomain.com" could be translated to "sip:*1*2345678900@gatewaydomain.com". Is there any built in functionality I can configure for this, or will this need to be scripted? Thanks, Ed

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[OT] the role of SBCs
by Daniel-Constantin Mierla 26 Oct '12

26 Oct '12

Hello, based on the outcome of the discussion carried in the thread: http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html I am looking again the clarify some aspects out there in the VoIP world. So, as mainly dealing with proxy/sip signaling deployments, it's very often to be the first one hit by support issues claiming things don't work. Then you investigate and end up in conclusions like in the above thread: "The problem was at the SBC, __where I did not expect it__." The underlined part heats me up a bit, because I never understood from where it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never mistaken or breaking things) node in a VoIP networks. To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is there to break the things. I don't want to start like a flame war, but is it something that I am obviously missing in regards to what benefits a SBC can bring? I see only inconveniences: - another point of failure - it is a b2bua, therefore very unlikely to offer the same performances of a proxy - if transcoding is needed, a media server can be used behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - if topology hinding is wanted in the b2bua fashion (not the proxy fashion with encoding headers), then the b2bua can be behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - nat traversal was solved long time ago in proxy environment, being scalable by deploying a farm of rtp proxy I don't want to go to other features, including the transport layer, it's a clear win of the proxy in my experience (ok, being deep involved in this project). Then, what makes the SBC so desirable in many companies/voip deployments? If any SBC user here that can share, what was the reason to buy such a device? Any conceptual functionality that cannot be achieved with the proxy as the first hop in front of the (wild) clients? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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MSRP: Generation of REPORTs on the relay
by Peter Dunkley 26 Oct '12

26 Oct '12

Hello, One of my colleagues at Crocodile has been doing some testing with the msrp module. With the auth change that was made yesterday, and the updated configuration I put in the README example it seems to mostly work. However, there is one set of scenarios relating to failure REPORTs that don't seem to be possible with the current module. RFC 4976 section 6.4.1 paragraph 3 states: If the Failure-Report header is "yes" or "partial", and if there is a problem processing the SEND request or if an error response is received for that SEND request, then the relay MUST respond with an appropriate error response in a REPORT back to the original source of the message. This means that when a SEND request that contains a Failure-Report: header with value of "yes" or "partial" is relayed by Kamailio we need to maintain some state so that when the TCP send fails, or we receive a failure response from the next hop, Kamailio can generate and send a REPORT. At the moment I don't think the msrp module has these capabilities. If the module is getting some re-work before the next release it would be good to get this in there. I am quite happy to help out in any way I can, but I am not sure of the best way to proceed with this. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

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Kamailio on Twitter: @kamailioproject
by Olle E. Johansson 26 Oct '12

26 Oct '12

The marketing arm of the Kamailio project has without further discussion opened up a Twitter account for Kamailio, to support our web site news, G+ and Facebook pages. Feel free to contribute with case stories and news we can publish on the web site and follow up in the social media flow. The twitter account is @kamailioproject - please follow it. And yes, Kamailio is THE best thing since the invention of sliced bread! This week's EXTRAORDINARY offer: Download TWO licenses of Kamailio for the same licensing cost as we've been offering for ONE license! And get an unlimited number of user licenses for your Kamailio SIP accounts! Kamailio greetings! Have a great SIP weekend! /Marketing

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MSRP relay authentication
by Peter Dunkley 25 Oct '12

25 Oct '12

Hi, MSRP AUTH requests must be authenticated using HTTP Digest authentication. Part of the concatenated data when doing this authentication is the method name. Because of the way MSRP requests are formatted, the part of the request-line that would contain the method name in HTTP or SIP requests is always MSRP. The MSRP method name (AUTH in this case) is at the end of the request-line. When Kamailio converts an MSRP request to SIP it has a method name of MSRP. This means that when Kamailio authenticates an MSRP request it uses "MSRP" as the method name, not the actual MSRP method name (AUTH). Is this correct? Should MSRP requests be authenticated with a method name of "MSRP" (which kind-of makes sense as this is the string at the start of the MSRP request line - which is where the method names are in HTTP and SIP) or should the MSRP method name of "AUTH" be used? If it is the later, does anyone know of an easy way to add this to Kamailio? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

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sr-users October 2012