Want to simplify Asterisk SIP message flows -
i trying implement sip ua , so, studied asterisk console in debug mode sip. tried call 1 extension (a) extension (b).
the initial message flow ringing message, understood, including digest authentication part. these are:
(b) >--->invite-----[asterisk]
followed series of back-and-forth messages:
401 unauthorized nonce ack invite correct digest trying trying ringing
after ringing phone (a) picked up, see following exchange of message:
(a) >----> ok >-----> [asterisk] (a) <----< ack<-----< [asterisk] [asterisk] >----- ok ------> (b) (a) >---(re)invite--> [asterisk] [asterisk] <-----ack-------< (b) [asterisk] >---(re)invite--> (b) (a) >---trying -----> [asterisk] [asterisk] <-----ok--------< (b) (a) >-----ok--------> [asterisk] (a) <----ack--------< [asterisk]
i writing ua part on (b) side , know sdp beforehand , can generate sdp b, in control. call flow b a. can control message going (b). how can reduce above message flow? also, not understand need many messages after initial sdps exchanged until ringing. or they?
asterisk described in sip standart(rfc).
you can't remove invites. things can do disable media , enable directmedia/ignore sdp part.
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