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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