Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
It is a temp solution. If your system continues growing up, you will have those problems again. Take a look at: http://wiki.freeswitch.org/wiki/Enterprise_Deployment You might find useful to install a second server. <http://wiki.freeswitch.org/wiki/Enterprise_Deployment>Regards 2010/12/7 Stephen Wilde <wstephen80@gmail.com>
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance,
Stephen
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There are a few performance tweaking tips at http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations. Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls. However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out. It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server). It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage. If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls. There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN. -Steve On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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Hi, 1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips? 2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview. thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference. There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists). -Steve On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2>The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible? On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service
providers
and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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It is, but it relies on the caller supporting 3xx. They might not handle the redirect. A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan. I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources. -Steve On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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The FS in front that doesn't handle the media is a good solution for me! The incoming SIP call can be bridged with a simple round robin rule to "FS media server" and a called prefix number can be added by front FS (and removed by FS media server) to propagate the "source IP address". Thank you for this info: I'll do some test on this solution. Stephen On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on
FS
box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a
server
from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on
FS
box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a
server
from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
Hi, I have one server running Freeswitch with some ISDN connections (via FreeTDM+Sangoma boards) and some SIP connections with service providers and customer. The usage of Freeswitch is as switching so it "bridge" each incoming call to a new outgoing call. SIP calls use G.729 and ISDN calls use ALaw for voice encoding. Now the number of call is grow up and also the CPU load is a little high so I have the necessity to scale UP my Freeswitch to handle more calls: what is the best way to do that? My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be this a solution? There are different way to scale UP? Thanks in advance, Stephen
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You can use X-Auth-IP with a FS-FS call too: Customer --> FS1 --> FS2 FS1 = front FS FS2 = media server 1. Create a proxy ACL on FS2 2. Add the IP of FS1 to that ACL 3. On FS1 do this in the dialplan: <extension ...> <condition ...> <action application="set" data="sip_h_X-Auth-IP=${network_addr}" /> <action application="bridge" data="sofia/gateway/fs2/..." /> </condition> </extension> FS2 will then be able to use the customer's IP in ACLs, user directory, etc. Remember to either set inbound_bypass_media=true on the sip profile, or <action application="set" data="bypass_media=true" /> in dialplan before the bridge. -Steve On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote: > Hi, > I have one server running Freeswitch with some ISDN connections > (via > FreeTDM+Sangoma boards) and some SIP connections with service > providers > and > customer. > The usage of Freeswitch is as switching so it "bridge" each > incoming > call to > a new outgoing call. > SIP calls use G.729 and ISDN calls use ALaw for voice encoding. > Now the number of call is grow up and also the CPU load is a > little > high > so > I have the necessity to scale UP my Freeswitch to handle more > calls: > what is > the best way to do that? > My first idea is to use a Sangoma D500 board to reduce the CPU > load. > Can > be > this a solution? > There are different way to scale UP? > Thanks in advance, > Stephen > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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Can someone explain what the difference is between using FS with bypass media and opensips? I've heard that opensips can handle a much higher CPS. -Avi On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <steveayre@gmail.com> wrote:
You can use X-Auth-IP with a FS-FS call too:
Customer --> FS1 --> FS2 FS1 = front FS FS2 = media server
1. Create a proxy ACL on FS2 2. Add the IP of FS1 to that ACL 3. On FS1 do this in the dialplan:
<extension ...> <condition ...> <action application="set" data="sip_h_X-Auth-IP=${network_addr}" /> <action application="bridge" data="sofia/gateway/fs2/..." /> </condition> </extension>
FS2 will then be able to use the customer's IP in ACLs, user directory, etc.
Remember to either set inbound_bypass_media=true on the sip profile, or <action application="set" data="bypass_media=true" /> in dialplan before the bridge.
-Steve
On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com>
wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing
on
FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote: > > There are a few performance tweaking tips at > > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations. > > Yes a Sangoma card will reduce your CPU load since transcoding won't > be done on the CPU any longer, that will then mean there's more CPU > available so you'll be able to handle more calls. > > However, if you're looking to increase your number of calls then you > probably want a cluster of servers as Juan pointed out. > > It'll mean you can increase the capacity by adding extra servers, so > there'd no longer be a limit to the number of calls you could handle > (just add another server). > > It'll also make maintenance easier, as you'll be able to pull a > server > from service for updates etc while traffic continues to run on the > other servers. Maintenance won't mean a service outage. > > If you're handling that many calls then additional servers would > make > your service more reliable. If a server crashes you'll still have > the > calls running on the other servers while you're fixing the problem > so > you won't have a complete outage. If FS is behind a load balancer > then > your customers might not even notice anything apart from a few > dropped > calls. > > There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will > attempt to continue calls if FS crashes and restarts, but I think > that's only for SIP-SIP not SIP-ISDN. > > -Steve > > > > > On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> > wrote: > > Hi, > > I have one server running Freeswitch with some ISDN connections > > (via > > FreeTDM+Sangoma boards) and some SIP connections with service > > providers > > and > > customer. > > The usage of Freeswitch is as switching so it "bridge" each > > incoming > > call to > > a new outgoing call. > > SIP calls use G.729 and ISDN calls use ALaw for voice encoding. > > Now the number of call is grow up and also the CPU load is a > > little > > high > > so > > I have the necessity to scale UP my Freeswitch to handle more > > calls: > > what is > > the best way to do that? > > My first idea is to use a Sangoma D500 board to reduce the CPU > > load. > > Can > > be > > this a solution? > > There are different way to scale UP? > > Thanks in advance, > > Stephen > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
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Well OpenSIPS is a proxy and FreeSWITCH is a B2BUA. If you just need to load-balance calls to your main FS boxes (which are used as B2BUA), to use another B2BUA for that task is not really useful. A SIP Proxy is generally more efficient for this, as they are dedicated to handle lots of REGISTER & INVITE, but if you want to avoid the trouble to deal with OpenSIPS config, which I find far less intuitive than FreeSWITCH, you may use FreeSWITCH in bypass-media as proposed in this thread. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 13/12/2010 à 14:26, Avi Marcus a écrit :
Can someone explain what the difference is between using FS with bypass media and opensips? I've heard that opensips can handle a much higher CPS. -Avi
On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <steveayre@gmail.com> wrote: You can use X-Auth-IP with a FS-FS call too:
Customer --> FS1 --> FS2 FS1 = front FS FS2 = media server
1. Create a proxy ACL on FS2 2. Add the IP of FS1 to that ACL 3. On FS1 do this in the dialplan:
<extension ...> <condition ...> <action application="set" data="sip_h_X-Auth-IP=${network_addr}" /> <action application="bridge" data="sofia/gateway/fs2/..." /> </condition> </extension>
FS2 will then be able to use the customer's IP in ACLs, user directory, etc.
Remember to either set inbound_bypass_media=true on the sip profile, or <action application="set" data="bypass_media=true" /> in dialplan before the bridge.
-Steve
On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote: > > There are a few performance tweaking tips at > > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations. > > Yes a Sangoma card will reduce your CPU load since transcoding won't > be done on the CPU any longer, that will then mean there's more CPU > available so you'll be able to handle more calls. > > However, if you're looking to increase your number of calls then you > probably want a cluster of servers as Juan pointed out. > > It'll mean you can increase the capacity by adding extra servers, so > there'd no longer be a limit to the number of calls you could handle > (just add another server). > > It'll also make maintenance easier, as you'll be able to pull a > server > from service for updates etc while traffic continues to run on the > other servers. Maintenance won't mean a service outage. > > If you're handling that many calls then additional servers would > make > your service more reliable. If a server crashes you'll still have > the > calls running on the other servers while you're fixing the problem > so > you won't have a complete outage. If FS is behind a load balancer > then > your customers might not even notice anything apart from a few > dropped > calls. > > There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will > attempt to continue calls if FS crashes and restarts, but I think > that's only for SIP-SIP not SIP-ISDN. > > -Steve > > > > > On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> > wrote: > > Hi, > > I have one server running Freeswitch with some ISDN connections > > (via > > FreeTDM+Sangoma boards) and some SIP connections with service > > providers > > and > > customer. > > The usage of Freeswitch is as switching so it "bridge" each > > incoming > > call to > > a new outgoing call. > > SIP calls use G.729 and ISDN calls use ALaw for voice encoding. > > Now the number of call is grow up and also the CPU load is a > > little > > high > > so > > I have the necessity to scale UP my Freeswitch to handle more > > calls: > > what is > > the best way to do that? > > My first idea is to use a Sangoma D500 board to reduce the CPU > > load. > > Can > > be > > this a solution? > > There are different way to scale UP? > > Thanks in advance, > > Stephen > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
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OpenSIPS / Kamailio / OpenSER is just a SIP Proxy. Proxy nearly in the same way than a HTTP proxy. It will normally just forward messages from A to B. OpenSIPS could also be used as a registrar. OpenSIPS does NEVER handle media- this task could be done within a scenario with mediaproxy/rtpproxy. FreeSWITCH with bypass_media enabled does not handle media but does still handle all the SIP requests. Simple example: OpenSIPS could forward a REGISTER to another peer without a check if the user does exists. FreeSWITCH would handle the REGISTER and all the other SIP messages. Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Avi Marcus Gesendet: Montag, 13. Dezember 2010 14:26 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Scale UP Freeswitch Can someone explain what the difference is between using FS with bypass media and opensips? I've heard that opensips can handle a much higher CPS. -Avi On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <steveayre@gmail.com> wrote: You can use X-Auth-IP with a FS-FS call too: Customer --> FS1 --> FS2 FS1 = front FS FS2 = media server 1. Create a proxy ACL on FS2 2. Add the IP of FS1 to that ACL 3. On FS1 do this in the dialplan: <extension ...> <condition ...> <action application="set" data="sip_h_X-Auth-IP=${network_addr}" /> <action application="bridge" data="sofia/gateway/fs2/..." /> </condition> </extension> FS2 will then be able to use the customer's IP in ACLs, user directory, etc. Remember to either set inbound_bypass_media=true on the sip profile, or <action application="set" data="bypass_media=true" /> in dialplan before the bridge. -Steve On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
There are a few performance tweaking tips at
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
Yes a Sangoma card will reduce your CPU load since transcoding won't be done on the CPU any longer, that will then mean there's more CPU available so you'll be able to handle more calls.
However, if you're looking to increase your number of calls then you probably want a cluster of servers as Juan pointed out.
It'll mean you can increase the capacity by adding extra servers, so there'd no longer be a limit to the number of calls you could handle (just add another server).
It'll also make maintenance easier, as you'll be able to pull a server from service for updates etc while traffic continues to run on the other servers. Maintenance won't mean a service outage.
If you're handling that many calls then additional servers would make your service more reliable. If a server crashes you'll still have the calls running on the other servers while you're fixing the problem so you won't have a complete outage. If FS is behind a load balancer then your customers might not even notice anything apart from a few dropped calls.
There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will attempt to continue calls if FS crashes and restarts, but I think that's only for SIP-SIP not SIP-ISDN.
-Steve
On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote: > Hi, > I have one server running Freeswitch with some ISDN connections > (via > FreeTDM+Sangoma boards) and some SIP connections with service > providers > and > customer. > The usage of Freeswitch is as switching so it "bridge" each > incoming > call to > a new outgoing call. > SIP calls use G.729 and ISDN calls use ALaw for voice encoding. > Now the number of call is grow up and also the CPU load is a > little > high > so > I have the necessity to scale UP my Freeswitch to handle more > calls: > what is > the best way to do that? > My first idea is to use a Sangoma D500 board to reduce the CPU > load. > Can > be > this a solution? > There are different way to scale UP? > Thanks in advance, > Stephen > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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Thanks Steve. I'll try it but as per other suggestion.. i'll try opensips in front. Regarding your example below, i have two concerns: 1. on FS2 (media FS) i am using xml_curl to authenticate the customer ip and then generate the bridge (depends on customer and called number etc..). So in that case i don't have ACL involved. FS2 also don't deal with sip registrations etc.. its used just for ip 2 ip communication. So i feel that if i send x-auth-ip to FS2 i can still use it and can follow my current implementation with xml_curl, right? but: 2. Even i use FS or opensips to inject xauth ip, and also use it on media FS to authenticate my original customer.. but what about if someone inject my real customer ip in xauth ip? that way anyone call send calls, right? On Mon, Dec 13, 2010 at 10:41 AM, Steven Ayre <steveayre@gmail.com> wrote:
You can use X-Auth-IP with a FS-FS call too:
Customer --> FS1 --> FS2 FS1 = front FS FS2 = media server
1. Create a proxy ACL on FS2 2. Add the IP of FS1 to that ACL 3. On FS1 do this in the dialplan:
<extension ...> <condition ...> <action application="set" data="sip_h_X-Auth-IP=${network_addr}" /> <action application="bridge" data="sofia/gateway/fs2/..." /> </condition> </extension>
FS2 will then be able to use the customer's IP in ACLs, user directory, etc.
Remember to either set inbound_bypass_media=true on the sip profile, or <action application="set" data="bypass_media=true" /> in dialplan before the bridge.
-Steve
On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com>
wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing
on
FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
1. i am thinking to use kamailo in front of FS boxes, is there any difference between kamailo and opensips?
2. if kamailo or opensips is running in front of FS, then will it send call to FS with original customer ip? so i can do billing etc on FS box -> actually i do IP based authentication and also ip based billing on FS box, so in case, i recieve kamailo ip on FS box then i'll loose the original customer overview.
thanks On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote: > > There are a few performance tweaking tips at > > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations. > > Yes a Sangoma card will reduce your CPU load since transcoding won't > be done on the CPU any longer, that will then mean there's more CPU > available so you'll be able to handle more calls. > > However, if you're looking to increase your number of calls then you > probably want a cluster of servers as Juan pointed out. > > It'll mean you can increase the capacity by adding extra servers, so > there'd no longer be a limit to the number of calls you could handle > (just add another server). > > It'll also make maintenance easier, as you'll be able to pull a > server > from service for updates etc while traffic continues to run on the > other servers. Maintenance won't mean a service outage. > > If you're handling that many calls then additional servers would > make > your service more reliable. If a server crashes you'll still have > the > calls running on the other servers while you're fixing the problem > so > you won't have a complete outage. If FS is behind a load balancer > then > your customers might not even notice anything apart from a few > dropped > calls. > > There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will > attempt to continue calls if FS crashes and restarts, but I think > that's only for SIP-SIP not SIP-ISDN. > > -Steve > > > > > On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> > wrote: > > Hi, > > I have one server running Freeswitch with some ISDN connections > > (via > > FreeTDM+Sangoma boards) and some SIP connections with service > > providers > > and > > customer. > > The usage of Freeswitch is as switching so it "bridge" each > > incoming > > call to > > a new outgoing call. > > SIP calls use G.729 and ISDN calls use ALaw for voice encoding. > > Now the number of call is grow up and also the CPU load is a > > little > > high > > so > > I have the necessity to scale UP my Freeswitch to handle more > > calls: > > what is > > the best way to do that? > > My first idea is to use a Sangoma D500 board to reduce the CPU > > load. > > Can > > be > > this a solution? > > There are different way to scale UP? > > Thanks in advance, > > Stephen > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
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1. on FS2 (media FS) i am using xml_curl to authenticate the customer ip and then generate the bridge (depends on customer and called number etc..). So in that case i don't have ACL involved. FS2 also don't deal with sip registrations etc.. its used just for ip 2 ip communication. So i feel that if i send x-auth-ip to FS2 i can still use it and can follow my current implementation with xml_curl, right? but:
Yes, you can still use it. You should be able to check both network_addr (which I believe will store the IP of FS1/OpenSIPS) and sip_h_X-Auth-IP variables. The sip_h_X-Auth-IP variable will contain the IP set in X-Auth-IP by FS1/OpenSIPS. It will be up to you in your application to check that network_addr is a trusted proxy and check X-Auth-IP against the list of customer IPs.
2. Even i use FS or opensips to inject xauth ip, and also use it on media FS to authenticate my original customer.. but what about if someone inject my real customer ip in xauth ip? that way anyone call send calls, right?
That is why you check the network_addr is coming from a trusted IP (FS1/OpenSIPS) (which mod_sofia does using the proxy ACL). You only allow that IP to set X-Auth-IP, and ignore that header if it comes from any other IP. - Steve On 13 December 2010 16:39, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve.
I'll try it but as per other suggestion.. i'll try opensips in front.
Regarding your example below, i have two concerns:
1. on FS2 (media FS) i am using xml_curl to authenticate the customer ip and then generate the bridge (depends on customer and called number etc..). So in that case i don't have ACL involved. FS2 also don't deal with sip registrations etc.. its used just for ip 2 ip communication. So i feel that if i send x-auth-ip to FS2 i can still use it and can follow my current implementation with xml_curl, right? but:
2. Even i use FS or opensips to inject xauth ip, and also use it on media FS to authenticate my original customer.. but what about if someone inject my real customer ip in xauth ip? that way anyone call send calls, right?
On Mon, Dec 13, 2010 at 10:41 AM, Steven Ayre <steveayre@gmail.com> wrote:
You can use X-Auth-IP with a FS-FS call too:
Customer --> FS1 --> FS2 FS1 = front FS FS2 = media server
1. Create a proxy ACL on FS2 2. Add the IP of FS1 to that ACL 3. On FS1 do this in the dialplan:
<extension ...> <condition ...> <action application="set" data="sip_h_X-Auth-IP=${network_addr}" /> <action application="bridge" data="sofia/gateway/fs2/..." /> </condition> </extension>
FS2 will then be able to use the customer's IP in ACLs, user directory, etc.
Remember to either set inbound_bypass_media=true on the sip profile, or <action application="set" data="bypass_media=true" /> in dialplan before the bridge.
-Steve
On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
hmmm... so doing that will also require X-Auth-IP, right or something more tricky can be done? On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre@gmail.com> wrote:
It is, but it relies on the caller supporting 3xx. They might not handle the redirect.
A lot won't because you could redirect them to anywhere, so lots of implementations will ignore the 3xx. FreeSWITCH for instance can either ignore a 3xx or will send the call back into the dialplan.
I think you'll have more success having a FS server in front of the others and bridging the call through to each server. If you set inbound_bypass_media=true on the SIP profile, the RTP media will bypass that server and go directly between the caller and the other FS box. That means that the call won't be using any CPU since it'll only wake up when a SIP packet is being sent/received. You'll still be creating a session through so it'll still be allocating memory to the call, a SIP proxy would use fewer resources.
-Steve
On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me. Since we are talking about HA options... Is it practically doable use it:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2 The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. ...possible?
On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
> 1. i am thinking to use kamailo in front of FS boxes, is there any > difference between kamailo and opensips?
They're both forks of OpenSER so for the most part there's little difference.
There are some small differences though since the fork. For example, opensips has a load_balancer module which kamalio does not (kamalio can still do load balancing but has a different interface to do so).
> 2. if kamailo or opensips is running in front of FS, then will it > send > call > to FS with original customer ip? so i can do billing etc on FS box > -> actually i do IP based authentication and also ip based billing > on > FS > box, so in case, i recieve kamailo ip on FS box then i'll loose > the > original > customer overview.
It will appear coming from the proxy IP. But there is a workaround. Configure a proxy ACL on the SIP profile and add your proxy IP to it. Then adjust your proxy routing rules so that it adds a X-Auth-IP header that contains the original IP. Anything coming from anything in the proxy ACL is trusted and FS will use the value from X-Auth-IP (if it exists).
-Steve
On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote: > Hi, > > 1. i am thinking to use kamailo in front of FS boxes, is there any > difference between kamailo and opensips? > > 2. if kamailo or opensips is running in front of FS, then will it > send > call > to FS with original customer ip? so i can do billing etc on FS box > -> actually i do IP based authentication and also ip based billing > on > FS > box, so in case, i recieve kamailo ip on FS box then i'll loose > the > original > customer overview. > > thanks > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> > wrote: >> >> There are a few performance tweaking tips at >> >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations. >> >> Yes a Sangoma card will reduce your CPU load since transcoding >> won't >> be done on the CPU any longer, that will then mean there's more >> CPU >> available so you'll be able to handle more calls. >> >> However, if you're looking to increase your number of calls then >> you >> probably want a cluster of servers as Juan pointed out. >> >> It'll mean you can increase the capacity by adding extra servers, >> so >> there'd no longer be a limit to the number of calls you could >> handle >> (just add another server). >> >> It'll also make maintenance easier, as you'll be able to pull a >> server >> from service for updates etc while traffic continues to run on >> the >> other servers. Maintenance won't mean a service outage. >> >> If you're handling that many calls then additional servers would >> make >> your service more reliable. If a server crashes you'll still have >> the >> calls running on the other servers while you're fixing the >> problem >> so >> you won't have a complete outage. If FS is behind a load balancer >> then >> your customers might not even notice anything apart from a few >> dropped >> calls. >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will >> attempt to continue calls if FS crashes and restarts, but I think >> that's only for SIP-SIP not SIP-ISDN. >> >> -Steve >> >> >> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> >> wrote: >> > Hi, >> > I have one server running Freeswitch with some ISDN connections >> > (via >> > FreeTDM+Sangoma boards) and some SIP connections with service >> > providers >> > and >> > customer. >> > The usage of Freeswitch is as switching so it "bridge" each >> > incoming >> > call to >> > a new outgoing call. >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding. >> > Now the number of call is grow up and also the CPU load is a >> > little >> > high >> > so >> > I have the necessity to scale UP my Freeswitch to handle more >> > calls: >> > what is >> > the best way to do that? >> > My first idea is to use a Sangoma D500 board to reduce the CPU >> > load. >> > Can >> > be >> > this a solution? >> > There are different way to scale UP? >> > Thanks in advance, >> > Stephen >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users@lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> > >> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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participants (7)
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Avi Marcus
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Bernhard Suttner
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David Ponzone
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Juan Antonio Ibañez Santorum
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Saeed Ahmed
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Stephen Wilde
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Steven Ayre