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
