[asterisk-ha-clustering] load balance register on two or more servers

Marco Mouta marco.mouta at gmail.com
Mon Nov 10 19:38:18 CST 2008


I'm not completely sure about this...

But I guess what's happening is:

1) you send the register request
2) your proxy forwards request to one of your asterisk A
3) you get the unauthorized with realm ( what's called a nonce or a
challenge from Asterisk A)
4) then you x-lite sends the answer for the challenge with your username and
password already encrypted md5...
5)your proxy doesn't keep the story as this is all udp so then he forwards
de MD5  register request answer to another asterisk B and then nothing will
work as this sip request is completely unknown this "new" asterisk getting a
register request already with answer to a md5 challenge...

I hope not getting you confused...

But i guess it can be the route cause of your problem.

kind regards,

--
Marco Mouta


On Mon, Sep 1, 2008 at 11:53 AM, ronald <nhadie at gmail.com> wrote:

> hi,
>
> i think i'm getting somewhere (i hope) with this combo.
>
> i have tried registering to the Virtual IP and i'm getting unauthorized.
>
> i set sip debug to try and see the difference and found out i am missing
> this:
>
> Authorization: Digest
> username="200200",realm="sip.mydomain.com",nonce="4cbc7dba",uri="sip:
> 123.45.67.130",response="76dafea9c97c5d94506d1249b7fdafad",algorithm=MD5
> Content-Length: 0
>
> when i try to register my phone using the virtual ip of the ldirectord
>
> but when i try to register using the actual ip address of the i can see
> that included on the REGISTER message. i am using x-lite.
>
> any clues why the Authorization Part is not there when i use the Virtual
> IP?
>
> TIA
>
> Regards,
> ronald
>
>
> ronald wrote:
> > Hi,
> >
> > I used ultramonkey on my web as well and got it working, now my prob is
> > the sip part:
> >
> > virtual=123.45.67.155:5060
> >          real=123.45.67.130:5060 gate
> >          real=123.45.67.131:5060 gate
> >          service=sip
> >          scheduler=rr
> >          protocol=udp
> >          checktype=negotiate
> >          persistent=1
> >
> > i defined that on ldirectord.cf, ipvsadm -L -n shows:
> >
> > Prot LocalAddress:Port Scheduler Flags
> >    -> RemoteAddress:Port           Forward Weight ActiveConn InActConn
> > TCP  123.45.67.155:443 rr persistent 600
> >    -> 123.45.67.131:443           Route   1      1          0
> >    -> 123.45.67.130:443           Route   1      0          0
> > UDP  123.45.67.155:5060 rr persistent 1
> > TCP  123.45.67.155:80 rr persistent 600
> >    -> 123.45.67.130:80            Route   1      0          0
> >    -> 123.45.67.131:80            Route   1      0          0
> >
> >
> > is there something wrong with my config? or i'm still missing something?
> >
> > regards,
> > ronald
> >
> >
> >
> > ronald wrote:
> >> i tried to follow this instruction:
> >>
> >>
> http://blog.iclutton.com/2008/01/load-balancing-and-high-availablity.html
> >>
> >> but i changed the config to
> >>
> >> virtual=103.222.18.155:5060
> >>         # first real server ip address, port, forwarding mecahnism,
> >> weight. masq -> NAT
> >>         real=103.222.18.130:5060 masq 1
> >>         # second real server ip address, port, forwarding mecahnism,
> >> weight. masq -> NAT
> >>         real=103.222.18.131:5060 masq 1
> >>
> >> then i tried my phone to register on 103.222.18.155
> >>
> >> i looked t the sniff i always get these:
> >>
> >> U 103.222.18.130:5060 -> 103.222.18.156:32770
> >>   SIP/2.0 404 Not Found..Via: SIP/2.0/UDP
> >> 103.222.18.156:32773
> ;branch=z9hG4bKhjhs8ass877;received=103.222.18.156..From:
> >> <sip:ldirectord at dev-server-2>;tag=
> >>   1928301774..To: <sip:ldirectord at dev-server-2
> >;tag=as58e3ee79..Call-ID:
> >> a84b4c76e66710..CSeq: 63104 OPTIONS..User-Agent: Asterisk PBX..Allow:
> >> INVITE, ACK
> >>   , CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported:
> >> replaces..Accept: application/sdp..Content-Length: 0....
> >> #
> >> U 103.222.18.156:32773 -> 103.222.18.131:5060
> >>   OPTIONS sip:ldirectord at dev-server-2 SIP/2.0..Via: SIP/2.0/UDP
> >> 103.222.18.156:32773;branch=z9hG4bKhjhs8ass877..Max-Forwards: 70..To:
> >> <sip:ldirectord at dev-
> >>   server-2>..From:
> >> <sip:ldirectord at dev-server-2>;tag=1928301774..Call-ID:
> >> a84b4c76e66710..CSeq: 63104 OPTIONS..Contact:
> >> <sip:ldirectord at dev-server-2>..Acc
> >>   ept: application/sdp..Content-Length: 0....
> >>
> >> #
> >> U 103.222.18.131:5060 -> 103.222.18.156:32770
> >>   SIP/2.0 404 Not Found..Via: SIP/2.0/UDP
> >> 103.222.18.156:32773
> ;branch=z9hG4bKhjhs8ass877;received=103.222.18.156..From:
> >> <sip:ldirectord at dev-server-2>;tag=
> >>   1928301774..To: <sip:ldirectord at dev-server-2
> >;tag=as2e6c3f7f..Call-ID:
> >> a84b4c76e66710..CSeq: 63104 OPTIONS..User-Agent: Asterisk PBX..Allow:
> >> INVITE, ACK
> >>   , CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported:
> >> replaces..Accept: application/sdp..Content-Length: 0....
> >> #
> >> U 103.222.18.156:32773 -> 103.222.18.130:5060
> >>   OPTIONS sip:ldirectord at dev-server-2 SIP/2.0..Via: SIP/2.0/UDP
> >> 103.222.18.156:32773;branch=z9hG4bKhjhs8ass877..Max-Forwards: 70..To:
> >> <sip:ldirectord at dev-
> >>   server-2>..From:
> >> <sip:ldirectord at dev-server-2>;tag=1928301774..Call-ID:
> >> a84b4c76e66710..CSeq: 63104 OPTIONS..Contact:
> >> <sip:ldirectord at dev-server-2>..Acc
> >>   ept: application/sdp..Content-Length: 0....
> >>
> >> #
> >>
> >> am i missing anything?
> >>
> >> regards,
> >> ronald
> >>
> >>
> >>
> >>
> >> ronald wrote:
> >>> Hi,
> >>>
> >>> What is the basic thing i need to load balance users registering on
> >>> the server?
> >>>
> >>> i have 2 asterisk using realtime, i just need to simply update my dns
> >>> not to use round-robin so i will point my domain to a single IP (i
> >>> know you are wondering y, but i made a mistake on using dns
> >>> round-robin on my web application and i'm losing session when a user
> >>> logs in because of it, so i have load balance apache using mod_proxy
> >>> now i need to load balance asterisk) i know it's confusing but for now
> >>> my goal is to load balance the registration of users, if i do load
> >>> balance on register, will invite be affected? or the load balancer
> >>> should just simply forward it to the asterisk?
> >>>
> >>> thank you
> >>>
> >>> regards
> >>> ronald
> >>>
> >>>
> >>> _______________________________________________
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> >>>
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> >>>
> >
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>
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