Roger Schreiter
2005-Feb-05 15:13 UTC
[Asterisk-Users] Problems with SIP invite due to long ping round trips
Hi, I'm installing asterisk 1.0.5 for a partner in China. Since the ping round trip takes typically 600 msec, I doubt, whether voice quality will we satisfying, but that is currently not my concern. The problem is, that most SIP phones or software (e.g. SJPhone) do resend the invite request, after approx 500 msec (measured by ethereal). chan_sip from asterisk seems to have a special handling for duplicate messages (i.e. messages with the same CSeq number) in order to ignore those messages in certain circumstances. Unfortunately asterisk doesn't ignore that second invite message by the client and sends an error message. (Though in the log files those "Too late messages" are mentioned.) Does anyone know a solution for the below shown scenery? Thanks for any hints! Roger. By the way: Registration succeeds without any problems. Following is the invite scenery: Client side <- approx 300 msec -> asterisk (Germany) (China P.R.) Invite -> (CSeq=1) -> Invite arrives (CSeq=1) <- 407: need authorization (CSeq=1) Resend Invite -> (CSeq=1) 407 arrives <- (CSeq=1) Ack CSeq=1 -> Invite with auth. -> (CSeq=2) -> Resent Invite arrives (without auth) (CSeq=1) <- 503: Not available (CSeq=1) -> Ack CSeq=1 arrives -> Invite with auth. arrives (CSeq=2) <- 407: need author. (CSeq=2) Resent Invite w.a. -> (CSeq=2) ... (Repeated approx 10 times than fails.)