<div dir="ltr">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.<div>-Avi<br><br><div class="gmail_quote">On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">You can use X-Auth-IP with a FS-FS call too:<br>
<br>
Customer --> FS1 --> FS2<br>
FS1 = front FS<br>
FS2 = media server<br>
<br>
1. Create a proxy ACL on FS2<br>
2. Add the IP of FS1 to that ACL<br>
3. On FS1 do this in the dialplan:<br>
<br>
<extension ...><br>
<condition ...><br>
<action application="set" data="sip_h_X-Auth-IP=${network_addr}" /><br>
<action application="bridge" data="sofia/gateway/fs2/..." /><br>
</condition><br>
</extension><br>
<br>
FS2 will then be able to use the customer's IP in ACLs, user directory, etc.<br>
<br>
Remember to either set inbound_bypass_media=true on the sip profile,<br>
or <action application="set" data="bypass_media=true" /> in dialplan<br>
before the bridge.<br>
<br>
-Steve<br>
<br>
<br>
<br>
On 12 December 2010 21:32, Saeed Ahmed <<a href="mailto:saeedahmad1981@gmail.com">saeedahmad1981@gmail.com</a>> wrote:<br>
> hmmm... so doing that will also require X-Auth-IP, right or something more<br>
> tricky can be done?<br>
> On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>> wrote:<br>
>><br>
>> It is, but it relies on the caller supporting 3xx. They might not<br>
>> handle the redirect.<br>
>><br>
>> A lot won't because you could redirect them to anywhere, so lots of<br>
>> implementations will ignore the 3xx. FreeSWITCH for instance can<br>
>> either ignore a 3xx or will send the call back into the dialplan.<br>
>><br>
>> I think you'll have more success having a FS server in front of the<br>
>> others and bridging the call through to each server. If you set<br>
>> inbound_bypass_media=true on the SIP profile, the RTP media will<br>
>> bypass that server and go directly between the caller and the other FS<br>
>> box. That means that the call won't be using any CPU since it'll only<br>
>> wake up when a SIP packet is being sent/received. You'll still be<br>
>> creating a session through so it'll still be allocating memory to the<br>
>> call, a SIP proxy would use fewer resources.<br>
>><br>
>> -Steve<br>
>><br>
>><br>
>> On 12 December 2010 19:28, Saeed Ahmed <<a href="mailto:saeedahmad1981@gmail.com">saeedahmad1981@gmail.com</a>> wrote:<br>
>> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.<br>
>> > Since we are talking about HA options... Is it practically doable use<br>
>> > it:<br>
>> ><br>
>> > <a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2" target="_blank">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2</a><br>
>> > The idea is to run one FS box (Redirect-FS) in front of several FS boxes<br>
>> > which redirect the call to active/available FS. If we make some script<br>
>> > on<br>
>> > redirect FS to count the active calls on media FSes and rearrange the<br>
>> > order<br>
>> > of redirect then loadbalacing can also be done.<br>
>> > ...possible?<br>
>> ><br>
>> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>><br>
>> > wrote:<br>
>> >><br>
>> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any<br>
>> >> > difference between kamailo and opensips?<br>
>> >><br>
>> >> They're both forks of OpenSER so for the most part there's little<br>
>> >> difference.<br>
>> >><br>
>> >> There are some small differences though since the fork. For example,<br>
>> >> opensips has a load_balancer module which kamalio does not (kamalio<br>
>> >> can still do load balancing but has a different interface to do so).<br>
>> >><br>
>> >> > 2. if kamailo or opensips is running in front of FS, then will it<br>
>> >> > send<br>
>> >> > call<br>
>> >> > to FS with original customer ip? so i can do billing etc on FS box<br>
>> >> > -> actually i do IP based authentication and also ip based billing on<br>
>> >> > FS<br>
>> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the<br>
>> >> > original<br>
>> >> > customer overview.<br>
>> >><br>
>> >> It will appear coming from the proxy IP. But there is a workaround.<br>
>> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.<br>
>> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP<br>
>> >> header that contains the original IP.<br>
>> >> Anything coming from anything in the proxy ACL is trusted and FS will<br>
>> >> use the value from X-Auth-IP (if it exists).<br>
>> >><br>
>> >> -Steve<br>
>> >><br>
>> >><br>
>> >><br>
>> >><br>
>> >> On 11 December 2010 14:00, Saeed Ahmed <<a href="mailto:saeedahmad1981@gmail.com">saeedahmad1981@gmail.com</a>><br>
>> >> wrote:<br>
>> >> > Hi,<br>
>> >> ><br>
>> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any<br>
>> >> > difference between kamailo and opensips?<br>
>> >> ><br>
>> >> > 2. if kamailo or opensips is running in front of FS, then will it<br>
>> >> > send<br>
>> >> > call<br>
>> >> > to FS with original customer ip? so i can do billing etc on FS box<br>
>> >> > -> actually i do IP based authentication and also ip based billing on<br>
>> >> > FS<br>
>> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the<br>
>> >> > original<br>
>> >> > customer overview.<br>
>> >> ><br>
>> >> > thanks<br>
>> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>><br>
>> >> > wrote:<br>
>> >> >><br>
>> >> >> There are a few performance tweaking tips at<br>
>> >> >><br>
>> >> >> <a href="http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations" target="_blank">http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations</a>.<br>
>> >> >><br>
>> >> >> Yes a Sangoma card will reduce your CPU load since transcoding won't<br>
>> >> >> be done on the CPU any longer, that will then mean there's more CPU<br>
>> >> >> available so you'll be able to handle more calls.<br>
>> >> >><br>
>> >> >> However, if you're looking to increase your number of calls then you<br>
>> >> >> probably want a cluster of servers as Juan pointed out.<br>
>> >> >><br>
>> >> >> It'll mean you can increase the capacity by adding extra servers, so<br>
>> >> >> there'd no longer be a limit to the number of calls you could handle<br>
>> >> >> (just add another server).<br>
>> >> >><br>
>> >> >> It'll also make maintenance easier, as you'll be able to pull a<br>
>> >> >> server<br>
>> >> >> from service for updates etc while traffic continues to run on the<br>
>> >> >> other servers. Maintenance won't mean a service outage.<br>
>> >> >><br>
>> >> >> If you're handling that many calls then additional servers would<br>
>> >> >> make<br>
>> >> >> your service more reliable. If a server crashes you'll still have<br>
>> >> >> the<br>
>> >> >> calls running on the other servers while you're fixing the problem<br>
>> >> >> so<br>
>> >> >> you won't have a complete outage. If FS is behind a load balancer<br>
>> >> >> then<br>
>> >> >> your customers might not even notice anything apart from a few<br>
>> >> >> dropped<br>
>> >> >> calls.<br>
>> >> >><br>
>> >> >> There's <a href="http://wiki.freeswitch.org/wiki/Freeswitch_HA" target="_blank">http://wiki.freeswitch.org/wiki/Freeswitch_HA</a> which will<br>
>> >> >> attempt to continue calls if FS crashes and restarts, but I think<br>
>> >> >> that's only for SIP-SIP not SIP-ISDN.<br>
>> >> >><br>
>> >> >> -Steve<br>
>> >> >><br>
>> >> >><br>
>> >> >><br>
>> >> >><br>
>> >> >> On 7 December 2010 12:26, Stephen Wilde <<a href="mailto:wstephen80@gmail.com">wstephen80@gmail.com</a>><br>
>> >> >> wrote:<br>
>> >> >> > Hi,<br>
>> >> >> > I have one server running Freeswitch with some ISDN connections<br>
>> >> >> > (via<br>
>> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service<br>
>> >> >> > providers<br>
>> >> >> > and<br>
>> >> >> > customer.<br>
>> >> >> > The usage of Freeswitch is as switching so it "bridge" each<br>
>> >> >> > incoming<br>
>> >> >> > call to<br>
>> >> >> > a new outgoing call.<br>
>> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.<br>
>> >> >> > Now the number of call is grow up and also the CPU load is a<br>
>> >> >> > little<br>
>> >> >> > high<br>
>> >> >> > so<br>
>> >> >> > I have the necessity to scale UP my Freeswitch to handle more<br>
>> >> >> > calls:<br>
>> >> >> > what is<br>
>> >> >> > the best way to do that?<br>
>> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU<br>
>> >> >> > load.<br>
>> >> >> > Can<br>
>> >> >> > be<br>
>> >> >> > this a solution?<br>
>> >> >> > There are different way to scale UP?<br>
>> >> >> > Thanks in advance,<br>
>> >> >> > Stephen<br>
>> >> >> ><br>
>> >> >> > _______________________________________________<br>
>> >> >> > FreeSWITCH-users mailing list<br>
>> >> >> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> >> >> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> >> >> ><br>
>> >> >> ><br>
>> >> >> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>> >> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> >> ><br>
>> >> >> ><br>
>> >> >><br>
>> >> >> _______________________________________________<br>
>> >> >> FreeSWITCH-users mailing list<br>
>> >> >> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> >> >> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> >> >><br>
>> >> >><br>
>> >> >> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>> >> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> ><br>
>> >> ><br>
>> >> > _______________________________________________<br>
>> >> > FreeSWITCH-users mailing list<br>
>> >> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> >> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> >> ><br>
>> >> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> ><br>
>> >> ><br>
>> >><br>
>> >> _______________________________________________<br>
>> >> FreeSWITCH-users mailing list<br>
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>> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> ><br>
>> ><br>
>> > _______________________________________________<br>
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>> ><br>
>> ><br>
>><br>
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><br>
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</blockquote></div><br></div></div>