sr-users March 2012

sr-users@lists.kamailio.org

87 participants
162 discussions

Best approach for TCP/TLS connection re-use for nated Contacts
by Giacomo Vacca 19 Mar '12

19 Mar '12

Hi all, I have a scenario where it's not clear to me what the best approach should be (Kamailio 3.2.0 on Squeeze, although I think it's a matter of routing logic only). I have a client behind NAT that sends an initial REGISTER request with a Contact containing private ip:port (e.g. Contact: "GV" <sip:12345@192.168.1.2:54321;transport=tcp> ) However, after it's challenged for authentication with a 401, it sends the following REGISTER request with Authorization header and with a Contact containing ip:port equal to the 'received' SIP URI (with a combination of 'rport' and 'received' parameters (e.g. Contact: "GV" <sip:12345@[receivedIP]:[rport];transport=tcp> ) In this case kamailio's routing logic is not marking the REGISTER as natted (no Received field in location and no NAT flags set). You can imagine I have a fairly standard NAT handling: route[NATDETECT] { force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } return; } FLT_NATS won't be set, and so it won't FLB_NATB in the following processing. This client receives a call: Kamailio relays the INVITE to it using the stored Contact. The INVITE is correctly received by the client, because the stored Contact refers to the connected socket ([received]:rport). The trouble starts when then the client replies with a 200 OK containing the private ip:port in the Contact header field (again 192.168.1.2:54321). The ACK will then have that private ip:port in the R-URI, and its relaying will fail. One of the solutions I've found is using always add_contact_alias() in onreply_route when handling the 200 OK, and then use handle_ruri_alias() when defining the destination for the ACK. I know that without traces and the full .cfg this is quite a broad question, but I wonder if you have already had to deal with this kind of client behaviour and can provide advice on the best practices. This is in particular useful for the re-use of TCP-based connections. Thanks in advance, Giacomo Truphone Limited is a limited liability company registered in England & Wales, registered office: 4 Royal Mint Court, London, EC3N 4HJ. Registered No. 04187081. VAT No. GB 851 5278 19. Tru is a brand name of Truphone and is a Truphone Communications Service. Truphone is a trading name for a number of distinct legal entities that operate in combination. www.truphone.com<http://www.truphone.com>. This e-mail, and any attachment(s), may contain information which is confidential and/or privileged, and is intended for the addressee only. If you are not the intended recipient, you may not use, disclose, copy or distribute this information in any manner whatsoever. If you have received this e-mail in error, please contact the sender immediately and delete it.

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register_udomain(): Invalid table version (use ser_mysql.sh reinstall)
by Ryan Gholam 19 Mar '12

19 Mar '12

Hello , I hope someone encountered this problem on UBUNTU , when connecting to the database i get this error in my log file :var/log/sys Mar 19 14:09:46 rayan-desktop /usr/sbin/ser[21807]: register_udomain(): Invalid table version (use ser_mysql.sh reinstall) Mar 19 14:09:46 rayan-desktop /usr/sbin/ser[21807]: domain_fixup(): Error while registering domain Any help is appreciated . Thank you .

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add_contact_alias and handle_ruri_alias vs fix_nated_contact
by yufei.tao 19 Mar '12

19 Mar '12

Hi I have been using fix_nated_register/fix_nated_contact for NAT traversal and all worked fine. I've come across add_contact_alias and handle_ruri_alias: http://sip-router.org/wiki/tutorials/alias-example where it says: "The benefits of using add_contact_alias() and handle_ruri_alias() functions instead of conventional NAT traversal solutions are: * Request-URI in requests to UAs behind NATs is always what UAs expect * Re-use of tcp/tls sessions between proxy and UAs " I assumed the 'conventional NAT traversal solution' here means fix_nated_contact? I understand the first point, in that the R-URI always contains what the client puts in the contact hf. But for the second point, about tcp/tls reuse, does fix_nated_contact do the same trick, or does add_contact_alias and handle_ruri_alias give you more benefit over fix_nated_contact? So far I feel they achieve the same thing but would like this confirmed/corrected. Thank you very much! Yufei

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Re: [SR-Users] dns lookup and store IP in avp..
by MingHon 19 Mar '12

19 Mar '12

On Mon, Mar 19, 2012 at 4:56 PM, MingHon <gminghon(a)gmail.com> wrote: > Thanks Daniel... :) > > > On Mon, Mar 19, 2012 at 4:18 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> >> On 3/19/12 8:22 AM, Klaus Darilion wrote: >> >>> No. If you really need it, then you can use exec() and a tool like "dig" >>> or "host". >>> >> apart of exec module functions, another option would be using an embedded >> language script (Lua/Perl...). >> >> Cheers, >> Daniel >> >>> >>> regards >>> Klaus >>> >>> On 19.03.2012 03:19, MingHon wrote: >>> >>>> Hi All, >>>> >>>> is this function available?? >>>> >>>> http://sourceforge.net/**tracker/?func=detail&aid=** >>>> 2687695&group_id=139143&atid=**743023<http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati…> >>>> <http://sourceforge.net/**tracker/?func=detail&aid=** >>>> 2687695&group_id=139143&atid=**743023<http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati…>> >>>> >>>> >>>> Thanks :) >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> MingHon >>>> >>> >>> ______________________________**_________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >> -- >> Daniel-Constantin Mierla >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.**php/kamailio-advanced-**training/<http://www.asipto.com/index.php/kamailio-advanced-training/> >> >> > > > -- > Regards, > > MingHon > -- Regards, MingHon

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Memory Question
by Klaus Darilion 19 Mar '12

19 Mar '12

Hi! Started kamailio with "-m 100 -M 30". Thus I would suspect that every process needs 30MB, and one of the processes needs 30+100Mb as it allocates the shared memory. The further would conclude that the memory will be allocated only once during startup. Can this be somehow verified? Normal "ps" output shows that every process needs ~130MB virtual memory, but this is probably not what I am looking for. Thanks Klaus # ps aux PID %CPU %MEM VSZ RSS TTY STAT COMMAND 846 0.2 0.8 139732 4112 ? S kamailio -m 100 -M 30 847 0.0 0.2 139732 1188 ? S kamailio -m 100 -M 30 848 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 849 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 850 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 851 0.0 0.2 139732 1220 ? S kamailio -m 100 -M 30 852 0.0 0.2 139732 1208 ? S kamailio -m 100 -M 30 853 0.0 0.2 139736 1032 ? S kamailio -m 100 -M 30 854 0.0 0.2 139732 1176 ? S kamailio -m 100 -M 30 855 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 856 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 857 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 858 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 859 0.0 0.2 139732 1396 ? S kamailio -m 100 -M 30

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Settings
by Robert Schleusener 19 Mar '12

19 Mar '12

Hello, on the iptel.org website I created an account. Also my own domain is registered and all works fine. I prefer numbers in the SIP-ID. With my own domain, I thought, I can use every number. But that is not true. Every number must start with the digit 8, for example 8555(a)mydomain.de. What is the reason for that? Have I the possibility to change that setting? I can also create aliases, for example 4123(a)mydomain.de and that works fine. But the telephonee sees still the number 8555 from the account name. Any ideas? Best regards, Robert

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dns lookup and store IP in avp..
by MingHon 19 Mar '12

19 Mar '12

Hi All, is this function available?? http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati… Thanks :) -- Regards, MingHon

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Hi Daniel.
by Andres Collazos 17 Mar '12

17 Mar '12

thanks for all the support for all this years. Can you please help me to know if there is any way to route sip calls based on transport protocol, for example a call incoming on tcp i will assign a route and if a call comes in udp i will assign a different route. my scenario is calls coming from different devices registering to kamailio and then kamailio send those calls to asterisk. unfortunately i have to create a different peer set for each device. for this scenario i have two types of UAs and they need completely different configuration on the asterisk switch. i am planing on segregate the traffic and build a media server for each type of device means 2 asterisk, teh only way that i can identify those incoming registrations is by the use of the port one client connects trough udp the other trough tcp. I appreciated any input in this matter. and again thank a lot to all for the great support. Andres Collazos.

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Presence in Kamailio
by Faisal Rehman 16 Mar '12

16 Mar '12

Hi, I am building Kamailio as a Stand-alone XMPP server so that XMPP clients can IM each other; but I also want the below functionality: The XMPP clients should all be able to see each others presence So do I need to look out the Kamailio presence module and will it fulfill my above requirement. Regards, Faisal Rehman

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Again problems with ACK and Bye
by Stoyan Mihaylov 16 Mar '12

16 Mar '12

In wireshark, there are 2 fields - Source and Destination. I am curious if there are variables in Kamailio which can show me those values. System: MYIP - IP of kamailio sip.mydomain.com - address of kamailio User (Imsdroid for Android) ->Kamailio->Asterisk->Kamailio (same) ->Other user. Problem - register, voice, video, and ring are OK, but ACK and BYE are going in dead loop - and this happens may be around in 50% of cases - not in all cases. If I am using other client (like hardware SIP phones), then I have no problems, but it is really hard to find some client for Android which supports video with Kamailio. To solve problem, I made one function in Kamailio.cfg - ACKBYE, which is called only within dialog. And within dialog is called in main route. route[ACKBYE] { #!ifdef WITH_MYFORWARD xlog("SCRIPT: AckBye $td;$si;$du;$dd\n"); if((($td=="sip.mydomain.com")||($td=="MYIP"))&&($si=="MYIP")){ $du=$sht(forw=>$ft); xlog("SCRIPT: AckBye2 $td;$si;$du\n"); return; } #!endif return; } Of cource I have if(is_method("INVITE")){ ds_select_dst("1","4"); $sht(forw=>$ft)=$du; } This worked for long time. But then we updated client (libraries etc), and problem appeared again - from time to time. Calls between 2 same clients can pass 2 or 5 times, and then problem appear. Before - $td and $si showed destination and source but now there is difference: ERROR: <script>: SCRIPT: AckBye 192.168.2.251;MYIP;<null>;<null> $td in this case show destination, which is correct, but in wireshark, destination and source = MYIP. ACK sip:MYIP;lr=on;nat=yes SIP/2.0 Via: SIP/2.0/UDP MYIP;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX ............................................ Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.2.105:44794;branch=z9hG4bK338581244;rport=44794 From: <sip:11171@sip.MYDOMAIN.com>;tag=1687441663 To: <sip:11175@192.168.2.251>;tag=as13c57096 Contact: <sip:11171@MYIP :5060;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: 4026cd88-aab9-19db-dfbe-1510baafdbb3 CSeq: 473985105 ACK Content-Length: 0 Max-Forwards: 0 Proxy-Authorization: Digest username="11171",realm="sip.stribogkonsult.com ",nonce="T2NZUU9jWCWxMUHSQweHXVQppHS8iFhd",uri="sip:11175@MYIP :5060",response="c5a95942af1502c208fa05d34c405907",algorithm=MD5 Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.0.453 (doubango r653)

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sr-users March 2012