It doesn't fix early media yet but brings us one step
closer to it:
The 183 (Session Progress) response is used to convey information
about the progress of the call that is not otherwise classified. The
Reason-Phrase, header fields, or message body MAY be used to convey
more details about the call progress.
Change-Id: Ibf264f251e41c06a7b4839acc0d0853e6400291c
In preparation of a better show calls VTY command it is of interest
to know which number has been dialed by whom. For that store the
source/dest in there.
MNCC: Change the talloc root context to the call and don't try to
free the strings after calling the routing code
SIP: Use talloc_strdup to duplicate them.
Call: Add null check because the talloc_strdup of the SIP layer
could have failed.
The codec negotiation is still a huge todo and the initial version
will be far from perfect. We will use whatever MNCC has decided on
and then see if it is compatible in the end.
Fix releasing of the leg in case it is not routable and make the
differentation if we initiated the invite (send CANCEL) or send
a final error. The error code was randomly picked and once we have
an enum of causes we can decide where to map it to.
Check if the SDP file has any codec potentially supported by GSM.
The topic of codec selection is a complicated one and we will not
support it correctly in the beginning.
* Create a new handle
* Send the invite
* Have some state transitions
* Allow to release a call in initial unconfirmed state, confirmed
one with cancel and connected with bye
* Add simple SDP parsing to find the rtpmap/codec that is used by
gsm
This code is capable of creating an agent that will bind on the
configured local address. The next steps are to configure the
library in terms of allowed features and prepare call handling.