[asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?

Johansson Olle E oej at edvina.net
Sat Jan 5 05:51:40 CST 2008


>>
>
> Olle--
>
> I wish you absolute best of luck in your search for funding. I'm
> painfully aware of how stressful such efforts can be.
Thanks.
>
>
> I'd love to discuss some of the options with you, if you get a free
> moment. Until then, I have a few questions:
>
> 1. on dialog matching-- if headers and tags are to be used for  
> matching
> multiple dialogs with the same callid, how often would this happen in
> the course of "normal" activity, and how many such dialogs (with the
> same callid) would normally exist; how many **COULD** exist? I'm going
> to try to read the sip spec so I can understand the whole dialog
> "thing". I'm not looking forward to it.

Well, if you just have a few SIP phones connected on a lan directly
to Asterisk, it will never happen.

If you have a stateful forking SIP proxy, it may happen in many
different ways. I'll try to write down a document of the scenarious
I've seen in testing. The most interesting way, and propably
something that happens now and then, is when a SIP proxy
forwards a call back to asterisk.

To give you some food for thought:

Let's say we have an outbound SIP call from Asterisk,
adressed to a SIP proxy. The SIP proxy forks back to
Asterisk for Voicemail after 30 secs so we have an
incoming call with the same call ID as an outbound
and it's not a loop. The other branch of the call goes
to a cell phone, so that branches back to Asterisk to
be forwarded over ZAP somewhere. Now we have
three SIP calls in Asterisk with the same call ID and
something that is very likely that it will happen.

Other than that, there are some theoretical scenarious that
may happen with non-stateful SIP proxys that we do test at
SIPits, but I haven't seen in any bug reports. THe case above
has been covered in bug reports.

When we redesign, we should try to fix it once and for all.
I'll start writing :-)

/O



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