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
