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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