Hi Ben, I agree with you. Though RFC3261 mandates "branch", BNF still permits and so parser will. Thanks for pointing out. Since SUBSCRIBE is a non-dialog creating method, I guess absence of branch doesn't make much difference apart from protocol compliance. This is just my thought.
Thanks and regards Nabam Serbang -----Original Message----- From: BONNAERENS Ben [mailto:[email protected]] Sent: Thursday, January 15, 2009 4:01 PM To: Serbang, Nabam (Nabam); Hasini Gunasinghe Cc: [email protected] Subject: RE: [Sip-implementors] SIP INVITE message with no branch parameterinits via header Hello, >With REGISTER, branch is not mandatory. This is incorrect. RFC3261 mandates the Via branch in every request. The predecessor of RFC3261 being RFC2543(bis) did not mandate the Via branch. The reason why software without Via branch still works is because RFC3261 mandates backward compatibility with RFC2543 implementations not having the Via branch (+ magic cookie) The Via branch + magic cookie should always be used in order to allow easier,faster transaction matching (RFC3261 chapter 17.2.3) Best regards, Ben. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Serbang, Nabam (Nabam) Sent: donderdag 15 januari 2009 11:21 To: Hasini Gunasinghe Cc: [email protected] Subject: Re: [Sip-implementors] SIP INVITE message with no branch parameterinits via header Hi Hasini, It will create prboblem when INVITE request is forked, med-dialog transaction involves such as answer-offer, reliable response, conference , transfer (most likely) etc. With REGISTER, branch is not mandatory. Thanks and regards Nabam Serbang -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Hasini Gunasinghe Sent: Thursday, January 15, 2009 3:34 PM To: [email protected] Subject: [Sip-implementors] SIP INVITE message with no branch parameter inits via header Hi all, I am implementing a SIP Soft Phone application using the RTC Client API. In both cases of initiating calls, that is; direct IP to IP and through asterisk SIP proxy, RTC Client does not include a branch parameter in its Via SIP header. But the calls are connected, media exchanged and calls are disconnected without any problem. My concern is, though RFC 3261says "The Via header field value MUST contain a branch parameter. This parameter is used to identify the transaction created by that request. This parameter is used by both the client and the server.", even without that, my app works fine so far. Can anybody please tell me when an error could occur if branch parameter is not present. And also is this parameter is taken in to consideration with lot of care in processing a REGISTER/INVITE requests? Thank you. regards, Hasini. _______________________________________________ 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 _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
