Re: [SR-Users] Kamailio with websocket as transport
Hello, On 20/03/15 01:48, Austin Einter wrote: Dear All I have a sip user agent. Today it works fine with Kamailio using tcp as tranasport. Now I am planning to support websocket in my sip ua. My question is does Kamailio support websocket transport. If so how can I configure it or while building kamailio do I need to make some flags on. I am using latest Kamailio 4.2.3 (version: kamailio 4.2.3 (x86_64/linux)). kamailio supports websocket since version 4.0. Look at the readme for websocket module to get started with it: - http://kamailio.org/docs/modules/stable/modules/websocket.html If you need to decrypt/encrypt SRTP, then see rtpengine module and rtpengine application. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio with websocket as transport
Dear All I have a sip user agent. Today it works fine with Kamailio using tcp as tranasport. Now I am planning to support websocket in my sip ua. My question is does Kamailio support websocket transport. If so how can I configure it or while building kamailio do I need to make some flags on. I am using latest Kamailio 4.2.3 (version: kamailio 4.2.3 (x86_64/linux)). Thanks Austin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Hardware config to run 1000 Kamailio TLS users
Hi All, 1. What are resource requirements for running Kamailio server with TLS module active for 1000 registered users (concurrent calls) when running on VPS? In other words, what amount of RAM and processor power required? 2. What is the maximum number of Kamailio TLS users/concurrent calls can be made for the below default values? # Amount of shared and private memory to allocate # for the running Kamailio server (in Mb) SHM_MEMORY=64 PKG_MEMORY=8 Thanks. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Hardware config to run 1000 Kamailio TLS users
Hello, On 19/03/15 15:44, elek...@yandex.ru wrote: Hi All, 1. What are resource requirements for running Kamailio server with TLS module active for 1000 registered users (concurrent calls) when running on VPS? In other words, what amount of RAM and processor power required? libssl need memory and CPU for doing encryption/decryption, so have at least 4 cores and allocate 256MB shared memory for kamailio. 2. What is the maximum number of Kamailio TLS users/concurrent calls can be made for the below default values? # Amount of shared and private memory to allocate # for the running Kamailio server (in Mb) SHM_MEMORY=64 PKG_MEMORY=8 It depends also on the other modules used in the config file. Perhaps you have to use sipp or other tool with your config to get some precise numbers. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] missing BYE when 2 redundant kamailio servers share the same database
Hello, On 19/03/15 02:54, Ding Ma wrote: [...] My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? can you get the pcap of such call? We have to see the routing headers to say what is next hop address. Are all the requests within dialog routed via same instance of kamailio? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio hardware requirements
Hello, another metric is how many tcp/tls connections you have to handle. Also, if you do caching or db only user location. For the proxies, a decent server these days (4-8 cores) with 2GB of shared memory and 12MB of private memory for kamailio should be enough. 16 to 32 children should be enough, more is no longer that efficient, either you have some slow components (e.g., database, dns), or too much traffic/users that you need another box. You may need to tune some of the parameters from usrloc (if you use caching) for better performances. Cheers, Daniel On 19/03/15 21:17, Mickael Marrache wrote: Hi, We are currently deploying an entire architecture composed of load balancers, proxies and media relays. All the components except the media relays are Kamailio instances. The media relays are RTP proxy instances. We are trying to determine the hardware requirements for the different servers. We will start with an architecture composed of: · 2 load balancers (one kept as backup to ensure availability) · 2 proxies (load is balanced over the 2 using the DISPATCHER module) · 2 media relays We are trying to determine the recommended hardware for the different components taking into account we will need to serve around 40 users. How many registrations per second can Kamailio support? How many concurrent calls can Kamailio support? What is the recommended number of children processes? What is the recommended size for PKG memory? What is the recommended size for SHM memory? Thanks for your help, Mickael ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Hardware config to run 1000 Kamailio TLS users
This is an interesting discussion. • on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections – consumed energy 209W http://www.kamailio.org/w/2011/05/green-voip-energy-efficiency-and-performaces-of-v3-0/ That paper by long-time contributor Jan Janak is a good one to return to now and then. /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamalio call issue
Hello, When we can change our DNS IP then it works with following : U 2015/03/20 10:06:06.504009 23.253.110.48:5060 - 202.157.76.21:64051 ACK sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48 SIP/2.0. Call-ID: 143610160aa023568302b9c999b79f45@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 23.253.110.48;branch=z9hG4bK7.601b877c92120368c17b3c0da0802af6.0. Via: SIP/2.0/UDP 192.168.0.217:5060;rport=63789;received=202.157.76.21;branch=z9hG4bK-353035- 5aa153400d9ce60dfc573738a4b55232. From: tester1 sip:tester1@23.253.110.48;tag=3fdb1d0f. To: tester2 sip:tester2@23.253.110.48;tag=31532119. Max-Forwards: 69. Contact: tester1 sip:tester1@192.168.0.217:5060;transport=udp;registering_acc=23_253_110_48; alias=202.157.76.21~63789~1. User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. But when we used another DNS IP (internet) and call then it showing only initialize.. If it has firewall issue then it will not work at all DNS IPs. I have attached before my config file for kamalio. Can you tell me what can be issue that it works at some DNS IP and not at all? Thanks From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, March 20, 2015 2:14 AM To: Yogendra Gupta; 'Kamailio (SER) - Users Mailing List' Subject: Re: [SR-Users] Kamalio call issue These are replies to INVITE requests, but if you see them, the INVITE passed through the server as well. If you are not aware of a firewall, then perhaps you don't have one unless is a default installation with it enabled or one on the network. I suggest you do sip tracing on the client machine to see if the invite requests leave to the proper IP. Ultimately can be also a problem caused by a NAT router with ALG, if the client is behind such device. Cheers, Daniel On 19/03/15 13:50, Yogendra Gupta wrote: Hello, When I am calling with other SIP user then I did not see any INVITE . that have issue with DNS. If we call with different DNS that is working fine then we see INVITE option like U 2015/03/19 12:39:01.744616 117.215.244.16:63380 - 23.253.110.48:5060 SIP/2.0 180 Ringing. CSeq: 2 INVITE. Call-ID: d83c4bc1e75e54df5ebd06b74f9089ef@0:0:0:0:0:0:0:0. From: tester1 sip:tester1@23.253.110.48 sip:tester1@23.253.110.48;tag=ef809ce0. To: sip:tester2@23.253.110.48 sip:tester2@23.253.110.48;tag=23a5eaea. Via: SIP/2.0/UDP 23.253.110.48;branch=z9hG4bKa1a3.21d8ef51bac2678fc26eca5975ae7b00.0,SIP/2.0/ UDP 192.168.0.217:5060;rport=62554;received=115.252.208.170;branch=z9hG4bK-34393 8-6f1017bcd9e693a4959717c9eabdc26e. Record-Route: sip:23.253.110.48;lr=on;nat=yes sip:23.253.110.48;lr=on;nat=yes. Contact: tester2 sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48 sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48 . User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. . U 2015/03/19 12:39:01.744870 23.253.110.48:5060 - 115.252.208.170:62554 SIP/2.0 180 Ringing. CSeq: 2 INVITE. Call-ID: d83c4bc1e75e54df5ebd06b74f9089ef@0:0:0:0:0:0:0:0. From: tester1 sip:tester1@23.253.110.48 sip:tester1@23.253.110.48;tag=ef809ce0. To: sip:tester2@23.253.110.48 sip:tester2@23.253.110.48;tag=23a5eaea. Via: SIP/2.0/UDP 192.168.0.217:5060;rport=62554;received=115.252.208.170;branch=z9hG4bK-34393 8-6f1017bcd9e693a4959717c9eabdc26e. Record-Route: sip:23.253.110.48;lr=on;nat=yes sip:23.253.110.48;lr=on;nat=yes. Contact: tester2 sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48; alias=117.215.244.16~63380~1 sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48; alias=117.215.244.16~63380~1. User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. Can you tell me what can be issue of firewall dropping? When I checked at server firewall: sudo ufw status Status: inactive Let me know what can be other issue for it.. Thanks From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Thursday, March 19, 2015 5:50 PM To: Yogendra Gupta; 'Kamailio (SER) - Users Mailing List' Subject: Re: [SR-Users] Kamalio call issue Hello, OPTIONS is not the request for initiating the calls, that is INVITE. You would need to know SIP a bit in order to be able to understand and configure Kamailio. If you don't see any INVITE on kamailo server via ngrep when you call, then the issue is on client side or there is a firewall dropping it. Cheers, Daniel On 19/03/15 11:39, Yogendra Gupta wrote: Hello, Thanks for nice support. When we call to test2 user and run this command at server ngrep -d any -qt -W byline sip port 5060 then we found following response at server: -- Daniel-Constantin Mierla http://twitter.com/#!/miconda http://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com -- Daniel-Constantin Mierla http://twitter.com/#!/miconda -
Re: [SR-Users] missing BYE when 2 redundant kamailio servers share the same database
We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to get a pcap anyway. Wonder if there is a way to dump pcap from inside kamailio. All the requests within dialog are routed through 2 kamailio instances. We want to make sure each phone only sends requests through its registrar. I have included pjsua logs in subsequent emails in this thread. Those logs have SIP messages, but only provide client perspective. Thanks for the help, Ding On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 19/03/15 02:54, Ding Ma wrote: [...] My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? can you get the pcap of such call? We have to see the routing headers to say what is next hop address. Are all the requests within dialog routed via same instance of kamailio? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com http://www.kamailioworld.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] missing BYE when 2 redundant kamailio servers share the same database
We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to get a pcap anyway. Wonder if there is a way to dump pcap from inside kamailio. Wireshark can decrypt SIP signalling sent over TLS connections if you provide server's private key to it. All the requests within dialog are routed through 2 kamailio instances. We want to make sure each phone only sends requests through its registrar. I have included pjsua logs in subsequent emails in this thread. Those logs have SIP messages, but only provide client perspective. Thanks for the help, Ding On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com wrote: Hello, On 19/03/15 02:54, Ding Ma wrote: [...] My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? can you get the pcap of such call? We have to see the routing headers to say what is next hop address. Are all the requests within dialog routed via same instance of kamailio? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany -http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamalio call issue
Hello, again, if the SIP messages don't get to kamailio box, then the problem is on client side, not on server side. If the request gets to kamailio and cannot do dns requests or they don't resolve, you will see a sip reply from kamailio. There is not an issue that can be revealed by the config of kamailio. On the client side, try to use tools like host, dig and see what happens with dns requests. Also, use there ngrep to see if there are sip packets sent out of that box. Cheers, Daniel On 20/03/15 11:20, Yogendra Gupta wrote: Hello, When we can change our DNS IP then it works with following : U 2015/03/20 10:06:06.504009 23.253.110.48:5060 - 202.157.76.21:64051 ACK sip:tester2@192.168.0.100:5060;transport=udp;registering_acc=23_253_110_48SIP/2.0. Call-ID: 143610160aa023568302b9c999b79f45@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 23.253.110.48;branch=z9hG4bK7.601b877c92120368c17b3c0da0802af6.0. Via: SIP/2.0/UDP 192.168.0.217:5060;rport=63789;received=202.157.76.21;branch=z9hG4bK-353035-5aa153400d9ce60dfc573738a4b55232. From: tester1 sip:tester1@23.253.110.48;tag=3fdb1d0f. To: tester2 sip:tester2@23.253.110.48;tag=31532119. Max-Forwards: 69. Contact: tester1 sip:tester1@192.168.0.217:5060;transport=udp;registering_acc=23_253_110_48;alias=202.157.76.21~63789~1. User-Agent: Jitsi2.6.5390Windows 7. Content-Length: 0. But when we used another DNS IP (internet) and call then it showing only initialize.. If it has firewall issue then it will not work at all DNS IPs. I have attached before my config file for kamalio. Can you tell me what can be issue that it works at some DNS IP and not at all? Thanks *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com] *Sent:* Friday, March 20, 2015 2:14 AM *To:* Yogendra Gupta; 'Kamailio (SER) - Users Mailing List' *Subject:* Re: [SR-Users] Kamalio call issue These are replies to INVITE requests, but if you see them, the INVITE passed through the server as well. If you are not aware of a firewall, then perhaps you don't have one unless is a default installation with it enabled or one on the network. I suggest you do sip tracing on the client machine to see if the invite requests leave to the proper IP. Ultimately can be also a problem caused by a NAT router with ALG, if the client is behind such device. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] missing BYE when 2 redundant kamailio servers share the same database
An alternative would be to test with tcp, from routing point of view should be the same if it is udp-tcp or udp-tls. Cheers, Daniel On 20/03/15 15:09, Vitaliy Aleksandrov wrote: We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to get a pcap anyway. Wonder if there is a way to dump pcap from inside kamailio. Wireshark can decrypt SIP signalling sent over TLS connections if you provide server's private key to it. All the requests within dialog are routed through 2 kamailio instances. We want to make sure each phone only sends requests through its registrar. I have included pjsua logs in subsequent emails in this thread. Those logs have SIP messages, but only provide client perspective. Thanks for the help, Ding On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com wrote: Hello, On 19/03/15 02:54, Ding Ma wrote: [...] My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? can you get the pcap of such call? We have to see the routing headers to say what is next hop address. Are all the requests within dialog routed via same instance of kamailio? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
