Re: [asterisk-users] SIP Listen Multiple Ports
Steve Edwards wrote: On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. On Mon, 4 Jan 2010, Vikram Ragukumar wrote: My customer is keen on using a hardware bridge to maximize throughput and also allow multiple servers. My boss is pressing me to maintain Kamailio and rtpproxy compatibility, and understand the tradeoffs in satisfying both. The link below has a diagram showing the way I'm going now: http://signalogic.com/images/openser_asterisk_sysconfig_dataflow.jpg Will the bridge preclude me from gracefully modifying my code to use more Kamailio functionality, if needed? I won't claim to have any insight into your system, but... Your diagram shows all SIP messages (unencrypted and decrypted) flowing through Kamailio. My guess is that you would have access to all Kamailio features. Steve, Thank you for your prompt response. If Kamailio is setup to listen on ports 5060 and 9090, port 9090 carries unknown SIP signaling information. Is it possible for Kamailio to dump these unrecognized signaling packets to a user space application which would process and return packets to Kamailio ? Would it be better to use libnetfilter_queue() to handle the unrecognized signaling information prior to Kamailio ? Thanks for all your help, Regards, Vikram. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
On Tue, 5 Jan 2010, Vikram Ragukumar wrote: If Kamailio is setup to listen on ports 5060 and 9090, port 9090 carries unknown SIP signaling information. Is it possible for Kamailio to dump these unrecognized signaling packets to a user space application which would process and return packets to Kamailio ? Would it be better to use libnetfilter_queue() to handle the unrecognized signaling information prior to Kamailio ? You're asking a blind man to describe an elephant. My knowledge in this area is so shallow you would be a fool to dive in based on anything I say :) I would be very surprised if Kamailio can deal with anything other that SIP, regardless of what port it is sent on. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
Hello, I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Olle E. Johansson wrote: No, Asterisk only supports one port. You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? If so, what would I be losing in not letting OpenSER do it? I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). Thanks, Vikram. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. If so, what would I be losing in not letting OpenSER do it? All the goodness that OpenSER brings to the table. I just scratched the surface above with all the features of OpenSER/Kamailio/OpenSIPS. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
Could you explain this one a bit more... You run openSER on the same box as asterisk, and have multiple such boxes, with the purpose of failover? But if a box goes down with openser on it, then there is no forwarding. (And most phones can only reg with peer). If you move openSER to another independent box, then you have a single point of failure. I suspect I'm not understanding something correctly -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, January 04, 2010 2:34 PM To: Asterisk Users List Cc: asterisk@sedwards.com Subject: Re: [asterisk-users] SIP Listen Multiple Ports 1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. If so, what would I be losing in not letting OpenSER do it? All the goodness that OpenSER brings to the table. I just scratched the surface above with all the features of OpenSER/Kamailio/OpenSIPS. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
Un-top-posting... On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Michelle Dupuis wrote: Could you explain this one a bit more... You run openSER on the same box as asterisk, and have multiple such boxes, with the purpose of failover? But if a box goes down with openser on it, then there is no forwarding. (And most phones can only reg with peer). If you move openSER to another independent box, then you have a single point of failure. I suspect I'm not understanding something correctly This is for a system advertised with those cheesy late night cable TV ads -- hot girl enticing you to call for a free chat and talk with all your friends... The hosts are located in a rural telco. The telco switch (Taqua 7000 as I remember) can hand off the call to a couple of IP addresses. Each of these addresses is on a separate host with OpenSER (it's been a few years) listening to 5060. Each of these hosts is also running Asterisk (1.2) listening on port 5061 There are no phones registering with any host. All calls come in through the Taqua or IAX when I need to test something. Each instance of OpenSER is configured (dispatcher.list) to distribute (no active load balancing) calls to all of the instances of Asterisk, including the instance running on the same host. If I want to take an instance of Asterisk down for maintenance, I just comment out the address associated with that instance out of dispatcher.list, restart the instances of OpenSER and wait for the in-progress calls to time out. If a host crashes, the Taqua detects that and doesn't send calls to that instance of OpenSER anymore. Each remaining instance of OpenSER will send calls to the remaining instances of Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
Hello, 1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. If so, what would I be losing in not letting OpenSER do it? All the goodness that OpenSER brings to the table. I just scratched the surface above with all the features of OpenSER/Kamailio/OpenSIPS. My customer is keen on using a hardware bridge to maximize throughput and also allow multiple servers. My boss is pressing me to maintain Kamailio and rtpproxy compatibility, and understand the tradeoffs in satisfying both. The link below has a diagram showing the way I'm going now: http://signalogic.com/images/openser_asterisk_sysconfig_dataflow.jpg Will the bridge preclude me from gracefully modifying my code to use more Kamailio functionality, if needed? Thanks and Regards, Vikram. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. On Mon, 4 Jan 2010, Vikram Ragukumar wrote: My customer is keen on using a hardware bridge to maximize throughput and also allow multiple servers. My boss is pressing me to maintain Kamailio and rtpproxy compatibility, and understand the tradeoffs in satisfying both. The link below has a diagram showing the way I'm going now: http://signalogic.com/images/openser_asterisk_sysconfig_dataflow.jpg Will the bridge preclude me from gracefully modifying my code to use more Kamailio functionality, if needed? I won't claim to have any insight into your system, but... Your diagram shows all SIP messages (unencrypted and decrypted) flowing through Kamailio. My guess is that you would have access to all Kamailio features. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time No, Asterisk only supports one port. /O like bindport = 5060,5061 OR bindport = 5060 bindport = 5090 I want, asterisk to listen SIP on multiple ports. so that users where SIP port 5060 blocked, can easily register to asterisk by using an alternate port. Shariq Khan On Fri, Jan 1, 2010 at 11:12 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Jan 1, 2010 at 10:34 AM, Shariq Khan shariqrazak...@gmail.com wrote: Is there any way to listen SIP on multiple ports on asterisk. Is is possible to define in sip.conf in the following way. sip.conf [general] port = 5060 port = 5090 Depending on the version of asterisk you are using, you can set the port that asterisk binds to using the following commands in sip.conf: 1.6.x: udpbindaddr = x.x.x.x:5061 1.4.x: bindport = 5061 -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Olle E. Johansson wrote: No, Asterisk only supports one port. You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
3 jan 2010 kl. 17.47 skrev Steve Edwards: 1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Olle E. Johansson wrote: No, Asterisk only supports one port. You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). That is a indeed a good piece of advice. With that solution, you can also properly support TCP and TLS on all versions of Asterisk. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
On Fri, Jan 1, 2010 at 10:34 AM, Shariq Khan shariqrazak...@gmail.comwrote: Is there any way to listen SIP on multiple ports on asterisk. Is is possible to define in sip.conf in the following way. sip.conf [general] port = 5060 port = 5090 Depending on the version of asterisk you are using, you can set the port that asterisk binds to using the following commands in sip.conf: 1.6.x: udpbindaddr = x.x.x.x:5061 1.4.x: bindport = 5061 -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Listen Multiple Ports
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time like bindport = 5060,5061 OR bindport = 5060 bindport = 5090 I want, asterisk to listen SIP on multiple ports. so that users where SIP port 5060 blocked, can easily register to asterisk by using an alternate port. Shariq Khan On Fri, Jan 1, 2010 at 11:12 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Jan 1, 2010 at 10:34 AM, Shariq Khan shariqrazak...@gmail.comwrote: Is there any way to listen SIP on multiple ports on asterisk. Is is possible to define in sip.conf in the following way. sip.conf [general] port = 5060 port = 5090 Depending on the version of asterisk you are using, you can set the port that asterisk binds to using the following commands in sip.conf: 1.6.x: udpbindaddr = x.x.x.x:5061 1.4.x: bindport = 5061 -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
