Version 1.12.12 of libsofia-sip-ua no longer automatically generates a
contact header element from the local ip address and port. Specifying
the contact tag does not break operation with the existing
library (1.12.11), but allows for operation on a system with
freeswitch 1.10.4 or later installed, which is built against this new
version of libsofia-sip-ua.
Sometimes, logging from sofia lacks the final newline character, messing up log
output. First snprintf() to a buffer, add '\n' if necessary and then log.
Up to now most logging is on LDEBUG, lets make more use of Log Levels.
reserve NOTICE for unusual events
INFO: normal call setup/teardown
DEBUG, well.. it's DEBUG
* BYE is not an Error.
* 4XX or 5XX response to INVITE is not an Error don't log as such.
* 183 does not necessarily mean "ringing".
Change those log messages for clarity.
Also removes a comment in sdp_create_file() about the
IP address in o= and c= having to be the same.
It is completely legal in SDP and often normal for the
originator and the connection information IP to be different.
Fixes a bug I introduced in 5f73c2033b
where we would not call mncc_call_leg_connect() on receiving 200 from
SIP side, and therefore never send MNCC_SETUP_RSP to the MS
Handle MO hold and retrieve and pass this to the SIP side.
Handle the 200 from the SIP side in response to our HOLD-ing re-INVITE.
With this commit we now handle MO hold and therefore also handle
call-waiting and swapping.
Add function pointers to the call_leg struct for call hold and retrieve.
Add function to send re-INVITE to SIP side when MNCC side puts call on HOLD/RETRIEVES.
Add MNCC/SIP CC_HOLD to call states.
SIP end points can send periodic re-INVITES. Previous to this commit,
the osmo-sip-connector would send a new call SETUP to the MSC for each
Add a function to find if we already handle this call based on the nua handle.
Use this function to detect and respond with an ACK to re-INVITES.
Add a function to extract the media mode from the SDP.
In the case the re-INVITE has a=sendonly (HOLD) respond with a=recvonly
In the case that the re-INVITE changes the media connection ip/port,
forward this to the MNCC side with an MNCC_RTP_CONNECT
This enables call hold implemented by subsequent commits
Prior to this commit, osmo-sip-connector would not send
any media mode attribute in the sdp. After this commit
we will by default always include a=sendrecv.
Given that a media mode attribute of "sendrecv" is default
and implicit it its absense, this does not represent any
Adds cause field to the call_leg and sip_call_leg structs.
Translates the SIP status to MNCC cause and vice versa and
uses this information in the SIP/MNCC messages at call leg
Implements a cause_map, mapping the SIP status codes
to GSM48_CC_CAUSE_* that are defined in libosmocore.
The map at the same time implements the Q.850 cause texts
that are subsequently used in the SIP Reason: header.
Implements two functions cause2status() and status2cause()
to read the map and return the the mapped status.
The mapping mostly follows the implemention in the LCR program,
so that any implementation relying on that mapping should
continue to work as expected with osmo-sip-connector.
sofia-sip allows applications to register a log backend function
which will be called every time the library wants to log something.
We register such a call-back and make it log using the libosmocore logging
The problem is that sofia-sip has its own log level management, and by
the time the message hits libosmocore, we don't know which log level we
shall use :(
Parse the media from session in progress and if present in alerting
connect the call early. Sadly this sets RTP to the sendrecv mode even
if we would like to keep it as recvonly.
So far the remote_port has never been used. sofia-sip did the right
thing and put the port into the "Contact" and the rport option for
the via. But we would have never been able to connect a PBX on a
different port (as sofia-sip seems to parse the destination from the
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.
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.
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.
* 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