Hello List, My call flow has initial INVITE and re-INVITE to update RTP IP/port. Usually everything works well, but sometimes OpenSIPS come up with following example:
UA OpenSIPS PSTN GW ------------------------------------------- INV(CSeq: 100) -----> | ---> INV(CSeq: 100) <---- 200 OK | <--- 200 OK (UA sends ACK then new INVITE) ACK(CSeq: 100) -----> | reINV(Cseq: 101) ---> | (OpenSIPS relays first INVITE then ACK) | ---> reINV(CSeq: 101) | ---> ACK(CSeq: 100) When PSTN gateway receives re-INVITE before ACK for previous INVITE it responds 500 with Retry-After header. This is correct behaviour which conforms to the RFC 3261 section 14.2 My question is: Is it possible to assure order of received and relayed messages within the same SIP session? Is there any configuration parameter? Thank you, -- Stas Kobzar Developeur VoIP / VoIP Developer Modulis.ca Inc. # Bureau / Office: 514-284-2020 x 246 Email: s <http://firstname.lastname>tas.kob...@modulis.ca https://www.modulis.com
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