Nothing that I have heard of!! By the way, even if you do, doesnt it sound like a hack? That's what I meant. The option a. is not really difficult.
+Fasih ----- Original Message ----- From: "Aswin Bhupalam" <[EMAIL PROTECTED]> To: "Fasih" <[EMAIL PROTECTED]>; <[email protected]> Sent: Thursday, April 06, 2006 9:03 AM Subject: Re: [Sip-implementors] SIP Server & Redundancy > Fasih, > Thank you very much for the mail. I appreciate your informative response. > > regarding response (b), can we not instruct DNS to make one IP active > through our application? > > Regards, > Aswin Bhupalam. > ----- Original Message ----- > From: "Fasih" <[EMAIL PROTECTED]> > To: <[email protected]> > Sent: Wednesday, April 05, 2006 10:48 PM > Subject: Re: [Sip-implementors] SIP Server & Redundancy > > >> Hi >> To me, the first option looks much better than the second, though I am > sure >> there will be people who would prefer the other way. >> Anyways, this is my bit: >> a. Using one virtual ip is pretty straight-forward. You do an ioctl with >> SIOCLIFADDIF >> Add a new logical interface on a physical interface >> using an unused logical unit number. >> >> SIOCLIFREMOVEIF >> Remove a logical interface by specifying its IP >> address or logical interface name.man if_tcp >> >> This sounds simple, all you have to do is to have the standby keep track > of >> the status of the primary to find out when it can take over the primary >> role. >> >> b. You can use multiple ip/ports in the same DNS record, that is simple > also >> [Though I have not done it, but I am very sure it is possible to define >> multiple ip/ports in the same recrds.] However, here the DNS server will > do >> the decission on who is going to be active. I dont think that it is that >> desirable. For e.g there might be some configuration problem on one > machine. >> Whereas it is reachable, it is not active. Or better still. You want to >> upgrade the s/w and you wud like to upgrade the secondary... ask it to >> become the primary.... bring down the primary when it is done with its >> processing... upgrade it..... bring it back to life.... now... if u use >> DNS.... I am not sure this would be possible without hacks..... >> >> Regards >> +Fasih >> >> > All, >> > How do we implement SIP Server redundancy in solaris? >> > a) Using One Virtual IP address(for SIP Signalling) for two different >> > solaris machines and >> > making it's interface as active/inactive depending on the status of >> > SIP Servers. >> > >> > b) Using One domain name and register the ipaddress of the Active SIP >> > Server with DNS. >> > >> > If we choose (a), how should we make the ethereal interface >> > active/inactive from the application. >> > Are there any APIs'/system-calls for the same? >> > >> > If we choose (b), how should we register the ipaddress of the active >> > SIP >> > Server to DNS from the application. >> > (The disadvantage here is, we should configure the domain name always >> > in >> > the SIP Phones). >> > >> > Also, please let me know which solution is better. >> > >> > Thanks & Regards, >> > Aswin Bhupalam. >> ----- Original Message ----- >> From: <[EMAIL PROTECTED]> >> To: <[email protected]> >> Sent: Wednesday, April 05, 2006 11:00 AM >> Subject: Sip-implementors Digest, Vol 37, Issue 9 >> >> >> > Send Sip-implementors mailing list submissions to >> > [email protected] >> > >> > To subscribe or unsubscribe via the World Wide Web, visit >> > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors >> > or, via email, send a message with subject or body 'help' to >> > [EMAIL PROTECTED] >> > >> > You can reach the person managing the list at >> > [EMAIL PROTECTED] >> > >> > When replying, please edit your Subject line so it is more specific >> > than "Re: Contents of Sip-implementors digest..." >> > >> > >> > Today's Topics: >> > >> > 1. Early media (Erez Morabia) >> > 2. SIP Server & Redundancy. (Aswin Bhupalam) >> > 3. Processing 301/302 with muliple contacts >> > (Srinivasa Rao Chigurupati) >> > 4. Re: Processing 301/302 with muliple contacts (Dale R. Worley) >> > 5. Query on PRACK (Rajendra, Subbu) >> > >> > >> > ---------------------------------------------------------------------- >> > >> > Message: 1 >> > Date: Wed, 5 Apr 2006 08:24:56 +0300 >> > From: "Erez Morabia" <[EMAIL PROTECTED]> >> > Subject: [Sip-implementors] Early media >> > To: <[email protected]> >> > Message-ID: >> > <[EMAIL PROTECTED]> >> > Content-Type: text/plain; charset="us-ascii" >> > >> > Hi, >> > >> > According toe RFC3264 (An Offer-Answer Model with SDP): >> > >> > 1. The IP address and port present in the offer indicate nothing about >> > the source IP address and source port of RTP and RTCP packets that will >> > be sent by the offerer. >> > >> > 2. Once the offerer has sent the offer, it MUST be prepared to receive >> > media for any recvonly streams described by that offer. It MUST be >> > prepared to send and receive media for any sendrecv streams in the >> > offer, and send media for any sendonly streams in the offer (of course, >> > it cannot actually send until the peer provides an answer with the >> > needed address and port information). In the case of RTP, even though >> > it >> > may receive media before the answer arrives, it will not be able to >> > send >> > RTCP receiver reports until the answer arrives. >> > >> > >> > >> > How can I protect my SIP UA against DOS attacks? How can I know what is >> > the real source IP/port of the RTP stream the sender sends toward my >> > SIP >> > UA? >> > >> > >> > >> > Thanks, >> > >> > Erez >> > >> > >> > >> > >> > >> > ------------------------------ >> > >> > Message: 2 >> > Date: Wed, 5 Apr 2006 11:42:15 +0530 >> > From: "Aswin Bhupalam" <[EMAIL PROTECTED]> >> > Subject: [Sip-implementors] SIP Server & Redundancy. >> > To: <[email protected]> >> > Message-ID: <[EMAIL PROTECTED]> >> > Content-Type: text/plain; charset="iso-8859-1" >> > >> > All, >> > How do we implement SIP Server redundancy in solaris? >> > a) Using One Virtual IP address(for SIP Signalling) for two different >> > solaris machines and >> > making it's interface as active/inactive depending on the status of >> > SIP Servers. >> > >> > b) Using One domain name and register the ipaddress of the Active SIP >> > Server with DNS. >> > >> > If we choose (a), how should we make the ethereal interface >> > active/inactive from the application. >> > Are there any APIs'/system-calls for the same? >> > >> > If we choose (b), how should we register the ipaddress of the active >> > SIP >> > Server to DNS from the application. >> > (The disadvantage here is, we should configure the domain name always >> > in >> > the SIP Phones). >> > >> > Also, please let me know which solution is better. >> > >> > Thanks & Regards, >> > Aswin Bhupalam. >> > >> > >> > ------------------------------ >> > >> > Message: 3 >> > Date: Wed, 05 Apr 2006 18:04:21 +0530 >> > From: Srinivasa Rao Chigurupati <[EMAIL PROTECTED]> >> > Subject: [Sip-implementors] Processing 301/302 with muliple contacts >> > To: [email protected] >> > Message-ID: <[EMAIL PROTECTED]> >> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed >> > >> > Hi >> > If UA gets 301/302 response to INVITE with multiple contacts >> > with/without q parameter, how should UA try new calls? >> > Should UA fork call to all contacts at once, or >> > should UA first try call to the contact with highest q value and say >> > that doesn't respond, should it try call to next contact? >> > >> > -- >> > Thanks & Regards >> > Srinivasa Rao Chigurupati >> > >> > >> > >> > >> > ------------------------------ >> > >> > Message: 4 >> > Date: Wed, 05 Apr 2006 09:18:22 -0400 >> > From: "Dale R. Worley" <[EMAIL PROTECTED]> >> > Subject: Re: [Sip-implementors] Processing 301/302 with muliple >> > contacts >> > To: Sip-Implementors <[email protected]> >> > Message-ID: <[EMAIL PROTECTED]> >> > Content-Type: text/plain >> > >> > On Wed, 2006-04-05 at 18:04 +0530, Srinivasa Rao Chigurupati wrote: >> >> If UA gets 301/302 response to INVITE with multiple contacts >> >> with/without q parameter, how should UA try new calls? >> >> Should UA fork call to all contacts at once, or >> >> should UA first try call to the contact with highest q value and say >> >> that doesn't respond, should it try call to next contact? >> > >> > RFC 3261 gives the UA-acting-as-a-proxy much latitude, but the >> > convention are: >> > >> > - contacts without q act as if they have q=1 >> > >> > - contacts are attempted in the groups that have the same q value, >> > highest q values first >> > >> > - contacts in a group are attempted in parallel >> > >> > Dale >> > >> > --- >> > interop.pingtel.com -- the public SIP phone interoperability test >> > server >> > >> > >> > >> > ------------------------------ >> > >> > Message: 5 >> > Date: Wed, 5 Apr 2006 11:54:26 -0400 >> > From: "Rajendra, Subbu" <[EMAIL PROTECTED]> >> > Subject: [Sip-implementors] Query on PRACK >> > To: "Sip-Implementors" <[email protected]> >> > Message-ID: >> > <[EMAIL PROTECTED]> >> > Content-Type: text/plain; charset="iso-8859-1" >> > >> > Hi, >> > If a UAS tries to send a provisional response reliably (by including >> > "Require: 100rel" header) to a UAC that does not support the 100rel >> > tag, >> > what should be the course of action that the UAC should take? Should >> > UAC >> > ignore this tag and process the provisional response unreliably or > should >> > it cancel the INVITE? Any pointers to this would be helpful. Thanks in >> > advance. >> > >> > Regards, >> > Subbu >> > >> > >> > >> > ------------------------------ >> > >> > _______________________________________________ >> > Sip-implementors mailing list >> > [email protected] >> > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors >> > >> > >> > End of Sip-implementors Digest, Vol 37, Issue 9 >> > *********************************************** >> > >> >> >> _______________________________________________ >> Sip-implementors mailing list >> [email protected] >> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
