sr-users August 2012

sr-users@lists.kamailio.org

77 participants
127 discussions

Question about Proxy-Authorization header in ACK
by Ivan Milivojevic 06 Aug '12

06 Aug '12

Hi all, I have a one question regarding ACK and Proxy-Authorization header. I am testing Kamailio 3.3 as SIP proxy,default config. I made one call with 2 phones where one has Public IP while other is behind NAT. SIP clients are Panasonic KX-UT133 and other is 1-NET (ex Sweden Mobile&CDMA provider ,over 50M users) Everything is working just fine, both RTP and SIP. On the other hand one of the clients,1-NET one, send Proxy-Authorization in ACK when he receives 200 OK from Kamailio. PUBLIC_IP_USER1 - user 1000 PUBLIC_IP_USER2 - user 1001 KAMAILIO ==> 1-NET SIP/2.0 200 OKr\n Record-Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes> KAMAILIO_PUBLIC_IP;lr=on;nat=yes> Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport=5064;branch=z9hG4bK934894606 Call-ID: 221374358 From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP> KAMAILIO_PUBLIC_IP>;tag=507511069 To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060> KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 CSeq: 21 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER Contact: <sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027> Require: replaces Content-Type: application/sdp Server: Panasonic_KX-UT133NE/01.081 (0080f0cedd83) Content-Length: 182 v=0 o=- 1343736535 1343736535 IN IP4 KAMAILIO_PUBLIC_IP s=- c=IN IP4 KAMAILIO_PUBLIC_IP t=0 0 m=audio 20412 RTP/AVP 18 a=rtpmap:18 G729/8000 a=sendrecv a=ptime:20 a=nortpproxy:yes 1-NET ==> KAMAILIO ACK sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027 SIP/2.0 Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport;branch=z9hG4bK642028490 Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes> KAMAILIO_PUBLIC_IP;lr=on;nat=yes> From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP> KAMAILIO_PUBLIC_IP>;tag=507511069 To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060> KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 Call-ID: 221374358 CSeq: 21 ACK Contact: <sip:1000@ <sip:1000@PUBLIC_IP_USER1:5064> PUBLIC_IP_USER1:5064> [truncated] Proxy-Authorization: Digest username="1000", realm="KAMAILIO_PUBLIC_IP", nonce="UBfL8lAXysb5tJCs80ZnthyPl9IzmRZk", uri="sip:1001@KAMAILIO_PUBLIC_IP:5060", response="7824519cdad9f1c2c79027a2d7522344", algorithm=MD5, cnonce="0a4f113b", q Max-Forwards: 70 User-Agent: Serbia_2.00 Content-Length: 0 I attach txt file with call flow, I can send pcap also. I think that the issue is related with bad client but I need another opinion. Does anyone has an idea about this issue? Best Regards, Ivan

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06 Aug '12

Hello, I'm trying to migrate an existing configuration from OpenSIPS 1.4.x to Kamailio 3.2.x. The old configuration replicates REGISTER requests to multiple servers using this code in the REGISTER routing block: add_sock_hdr("Local-Sock"); add_rcv_param(); append_branch("sip:1.1.1.1:5060"); t_replicate("sip:2.2.2.2:5060"); exit; From what I read in the documentation, the behaviour of append_branch() is different in Kamailio, so I modified my code like this: add_sock_hdr("Local-Sock"); add_rcv_param(); append_branch(); seturi(sip:1.1.1.1:5060"); t_replicate("sip:2.2.2.2:5060"); exit; Unfortunately, I do not see any REGISTER requests arriving on 1.1.1.1 so I assume that I made a mistake when forking the request. Can anyone point me in the right direction? Thanks, Henning Holtschneider -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 sip 25(a)voip.loca.net Registergericht Amtsgericht Dortmund HRA 14208 Geschäftsführer Sven Haufe, Henning Holtschneider

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Re: [SR-Users] sr-users Digest, Vol 87, Issue 9
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi Konstantin, Thanks for your kind reply. You can check the iptables output at : http://pastebin.com/i3zUfVeb. I hope that this will give you enough clue in right direction. With Regards, Vijay Thakur ====================================================================================================================== Message: 3 Date: Fri, 3 Aug 2012 17:38:23 +0300 From: "Konstantin M."<evilzluk(a)gmail.com> Subject: Re: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID: <CAEYtTzdeY88X-GC+wDwLou+smsB60h+M36h7xwCu448V0YtD2g(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Looks like your firewall is passing a state RELATED,ESTABLISHED and don't have a permit rule for a state NEW. Check your iptables or simply do a command: iptables-save > /tmp/iptables.txt and paste this file (/tmp/iptables.txt) to pastebin 2012/8/3 Vijay Thakur<vijay.thakur(a)loopmethods.com> > Hello all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. When > i make a call from Soft phone (X-Lite) to iphone, all is working fine. But > in other case call from iphone to Softphone is not working, even not > ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:**75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP > SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK > URGP=0 OPT (0101080A000E20610932B25A) > > This is server is hosted on Linnode. > > Kindly guide me to solve the problem > > -- > Best Regards, > > Vijay Thakur > (Assistant Manager - Networks) > Mobile : +91 8744018065 > Mail :vijay.thakur@loopmethods.com

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Gorka left a message for you
by Badoo 05 Aug '12

05 Aug '12

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Getting started with Kamailio
by Markus 05 Aug '12

05 Aug '12

Hello! I'm new to this list and Kamailio. First, thanks to all the programmers that developed this software. I'd like to ask several questions on how to get started with Kamailio and the basic understanding of it. I'm trying to figure out Kamailio only since yesterday. Right now I have a single Asterisk server that handles signaling and media. For accounting and user administration I'm using a2billing with MySQL on it. For redundancy and scalability I would like to create the following setup: France: BGP anycast 195.5.5.0/24 2 nodes with Kamailio (IP 195.5.5.5) active + standby 2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster. Spain: BGP anycast 195.5.5.0/24 2 nodes with Kamailio (IP 195.5.5.5) active + standby 2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster. (a2billing MySQL DB will be always synchronous over all nodes in Spain and France) This will give me: - Network redundancy (data center burns down in France, I stay online in Spain). This will be handled by BGP anycast. - Signaling redundancy (Kamailio active + standby). This will be handled by Linux-HA or something like that. - Media redundancy (2x Asterisk per country). This will be handled by Kamailio dispatcher. - Application/database redundancy (2x a2billing per country + synchronous MySQL DB everyhwere). This will be handled by DRBD and MySQL. = High redundancy, I can sleep and it can scale. What I would like to achieve besides the above: - Give only a single IP address to all customers and termination providers (the same IP address), 195.5.5.5. I'm a SIP noob, so I have to ask: - How do I do the Kamailio part? ;-) ... I have seen this how-to for Kamailio/Asterisk realtime: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb But it feels like "overkill". If I would not have any users that actually REGISTER (e.g. in a pure wholesale termination environment), I would not need Kamailio/Asterisk realtime integration. Correct? - Is there no way around changing the Asterisk side (activating realtime, new MySQL DB) when I have users that do need to REGISTER? If I would not be using a2billing I could probably handle all the registrations in Kamailio only? - Since Kamailio and Asterisk will not be on the same box, what is the recommended way for Kamailio securely communicating with the MySQL database on the Asterisk server? Does Kamailio support SSL with MySQL? - If I use RTPproxy on the Kamailio server, every customer and termination provider would connect to 1 single IP address, because both media and signaling comes from that IP. Correct? - If I don't use RTPproxy, and have canreinvite=yes on my Asterisk servers, customers would get the media, when placing PSTN calls, directly from my termination providers (I would like to avoid that). Correct? - If I don't use RTPproxy, and have canreinvite=no on my Asterisk servers, customers will get the media directly from my Asterisk servers, but termination providers that filter based on IP addresses they would have to allow all Asterisk IP addresses in their filters (same for customers, actually). Correct? Right now I didn't have to worry about such things because media and signaling were handled by a single Asterisk box with canreinvite=no everywhere. - Last question for now: why does it seem like important developers of SIP software such as Kamailio and yate are originating from Romania and are female? Just a coincidence? :-) Regards Markus

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RTP over TCP
by Techie Sup 05 Aug '12

05 Aug '12

Hi All, Is there any possibility of sending RTP over TCP, because we are building a softphone based on PJSIP, but seems that some ISP are blocking RTP packets over UDP. Any solutions for this. Regards Gloria

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kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error
by Vijay Thakur 03 Aug '12

03 Aug '12

Hello all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is server is hosted on Linnode. Kindly guide me to solve the problem -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

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302 redirect with 2 numbers registered
by Mino Haluz 03 Aug '12

03 Aug '12

Hi, one number is registered on 2 phones. Phone1 has Always redirect set to another number. When incoming call is initiated, Phone2 is ringing and Phone1 sends 302 to the proxy. However the proxy does not send 302 to the caller (for ex. GW), but it waits for timeout of the Phone2. Then the proxy sends 302 to the caller. Can I do in kamailio, that it will ring on the Phone1 and also on the number where it is redirected? I know kamailio is a proxy and cannot initiate a call, but is there any solution? Thanks. Mino

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Question on ACK behavior
by Varsha Venkatraramani 03 Aug '12

03 Aug '12

** ** * * ** ** Hi, ** ** I have a question regarding the below ACK response Kamilaio receives from a carrier. Can someone please help me understand why the “*Route: sip:callmanager@192.168.160.43:5060> *sent from Kamailio is missing a “<” as shown in the captures below? Our Internal proxy is treating that as a malformed header and dropping the packet. ** ** ** ** *ACK from CARRIER* * * U 2012/08/01 18:32:52.219852 4.55.18.227:5060 -> 192.168.160.47:5060**** *ACK sip:callmanager@192.168.160.43:5060 SIP/2.0.* Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.**** From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.**** To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.**** Call-ID: 295226_50030734(a)4.55.18.227.**** CSeq: 18079 ACK.**** Max-Forwards: 70.**** Route: <sip:192.168.160.47:5060;lr=on>.**** *Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060>.* Content-Length: 0.**** .**** *ACK FORWARDED TO SIP PROXY* ** ** U 2012/08/01 18:32:52.220612 192.168.160.47:5060 -> 192.168.160.44:5060**** *ACK sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060 SIP/2.0.* Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.**** Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.**** From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.**** To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.**** Call-ID: 295226_50030734(a)4.55.18.227.**** CSeq: 18079 ACK.**** Max-Forwards: 69.**** Content-Length: 0.**** *Route: sip:callmanager@192.168.160.43:5060>.* I have attached the config file for you reference. Kamailio version is 3.2.3 **** ** ** Thank You**** Varsha****

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centos (fedora) service startup file
by andre 02 Aug '12

02 Aug '12

Somwhere losted since openser 1.0 file kamailio.init Please fix in function start: start() { echo -n $"Starting $prog: " ..... [ $RETVAL = 0 ] && touch something to [ $RETVAL = 0 ] && touch someting && success I mean call function success in the end of the expression. Thanks in advance.

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sr-users August 2012