Bogdan Pintea wrote:
On 11/16/10 21:49, Stefan Sayer wrote:
I'm having some difficulties to understand the new architecture of
events processing. More specifically, I find it a bit strange to have
the AmSession::process() function call, through the AmSip*Event()
functions the AmSession::dlg::updateStatus() functions.
I think that this was the most effective code obfuscation possible... ;)
but the reason for it is, I think, that at some time in the future, the
AmSession should just be the *application* logic, and this should make
it possible to move a SIP dialog (=a SIP call) from one to another
application, or also possibly to manage several SIP calls with one app
logic instance.
The pain I have from this mis-understanding: with the 100rel, I need to
check some dialog sanity conditions (like 100rel is "required", but some
INVITE is missing support for it) and break the call (=the session) if
this is not met. For this, I'd need access AmSession::setStopped(), but
all I have is the API allowed by the AmSipDialogEventHandler.
can't you throw an AmSession::Exception? this should reply with the
error code and stop the session properly.
Mmm, well, thing is that if it's an INVITE, I need to stop the session.
If a re-INVITE, just to reject it (and keep the session).
Now, I could signal this in the Exception (smth. like 1=keep session,
0=kill it), but then I'd spread the 100rel logic in both classes, which
would be a hack.
no, the proper solution is to catch the exceptions when handling
re-invite separately, and just only reply to the request without
stopping the session.
Stefan
Well, the quick answer would be: put the 100rel stuff in AmSession.
I was just at this moment wondering about it. Shouldn't actually this
stuff be in AmSipDialog?
That's what I've started, but don't know how to do it "cleanly".
Well, for now, I could then extend the AmSipDialogEventHandler (with
session tearing down functionality), till the clear separation between
the two (AmSession <-> AmSipDialog) is delivered.
What do you think?
Bogdan.
Stefan
--
Stefan Sayer
VoIP Services Consulting and Development
Warschauer Str. 24
10243 Berlin
tel:+491621366449
sip:[email protected]
email/xmpp:[email protected]
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