Hi, all
I have two questions for route information preprocessing of initial INVITE with Route headers first question: UA(or B2BUA) receives an initial INVITE with Route headers, assuming UA's host is example.com if the sip entity which sends the INVITE is a loose router, the INVITE may be like this: INVITE sip:b...@example.com Route: <sip:x...@example.com;lr> if the sip entity which sends the INVITE is a strict router, the INVITE may be like this: INVITE sip:x...@example.com Route: <sip:b...@example.com> we cannot distinguish those two scenarioes because Requet-URIs in two scenarioes are similar. if we do route information preprocessing according RFC3261 section 16.4, we will get two different results. And there is no desciption for UA route information preprocessing in RFC3261, RFC3261 section 16.4 just describes proxy behavior. How to processing route information of initial INVITE with Route headers for UA? second question: as RFC3261 section 16.4 descibed, proxy must check whether the request-uri was placed by the proxy previously. But the request-uri in initial INVITE must not be placed by the proxy previously. Is something wrong with RFC3261 section 16.4? Best Regards LionEagle -------------------------------------------------------- ZTE Information Security Notice: The information contained in this mail is solely property of the sender's organization. This mail communication is confidential. Recipients named above are obligated to maintain secrecy and are not permitted to disclose the contents of this communication to others. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. Any views expressed in this message are those of the individual sender. This message has been scanned for viruses and Spam by ZTE Anti-Spam system. _______________________________________________ Sip mailing list https://www.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implement...@cs.columbia.edu for questions on current sip Use sipp...@ietf.org for new developments on the application of sip