[Asterisk-Users] Sip protocol question ...

[Robert Rozman]

Hi,

I'm tryinig to debug SIP call from activex control based on MS RTC (A) to
Asterisk (B). I use Etherreal to follow packages and I would like to ask
short questions:
- Session trace shows following order of packets:
        A - >  B    Invite
        B - >  A    100 Trying
        B - >  A    200 OK, with session description         ; repeated 6
times
        A - > B     BYE sip: ....
        B - > A     200 OK
- in my newbie logic it seems that B simply disconnects for some reason. In
session description there are codec specs. Unfortunately I don't have much
docs on this active x control, so don't know how it behaves or whether it
works.

But anyway, does B anyhow tells reason why it requests disconnection ?

Could I somehow from SIP packets gain knowledge about possible cause of
disconnection ?


Thanks in advance,

regards,

Robert.