* Remove unused RAII_VAR() declarations. The compiler cannot catch these
because the cleanup function "references" the unused variable. Some
actually allocated and released resources that were never used.
* Fixed some whitespace issues in stasis_bridges.c.
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The core may (depending on circumstances) request a single codec on outgoing
calls. Many channel drivers ignore or treat this as a suggestion while still
including configured codecs. The res_pjsip_session logic treated this as
an explicit request, leaving out other configured codecs.
This change makes res_pjsip_session behave like other channel driver and simply
adds the requested codec to the list.
(closes issue ASTERISK-23082)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3140/
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This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
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Channel creation in Asterisk is broken up into two steps: requesting and calling.
In some cases a channel may be requested but never called. This happens in the
ChanIsAvail dialplan application for determining if something is reachable or
not. The PJSIP channel driver did not take this situation into account and
attempted to end a session that was never called out on.
The code now checks the session state to determine if the session has been
called out on and if not terminates it instead of ending it.
(closes issue ASTERISK-23074)
Reported by: Kilburn
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The process for resending an INVITE with authentication involves restarting the UAC
session. We were incorrectly passing in that a new offer is being sent, causing the
SDP negotiation to get into a (technically speaking) funky state.
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The direct media format capabilities are always allocated in
ast_sip_session_alloc and were not freed in the session destructor. Whoops.
(This being the third whoops caught by Scott and Nitesh's valgrind work for
the Asterisk Test Suite. Nifty!)
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The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.
(closes issue ASTERISK-22668)
Reported by: John Bigelow
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Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
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Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for
the SDP to be negotiated *after* a channel was created and after it was being wait
on by an application. It is only after negotiation occurs that the file descriptors
for RTP are placed on the channel. Since the channel was already being waited on
these file descriptors were not monitored, causing incoming media to never be read.
This change wakes up any application waiting on the channel so that added file
descriptors end up being monitored.
(closes issue AST-1227)
Reported by: John Bigelow
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There was a collision of mod_data use on the transaction between using a nat
hook and an session response callback. During state change it was assumed
what was in the mod_data was nothing or the response callback. However, it
was possible for it to also contain a nat hook thus resulting in a bad cast
and a crash.
Added the ability to store multiple data elements in mod_data via a hash table.
In this instance, mod_data now stores a hash table of the two values that can
be retrieved using an associated string key.
(closes issue ASTERISK-22394)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2843/
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During load time in res_pjsip if an error occurred the operation would attempt to rollback all
operations done during load. This is not permitted by PJSIP as it will assert if the operation has
not been done. This fix changes the code so it will only rollback what has been initialized already.
Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to
limitations within PJSIP itself. The library environment can only be changed to a certain extent
and does not provide the ability, currently, to deinitialize certain required functionality.
(closes issue ASTERISK-22474)
Reported by: Corey Farrell
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RFC 5407 section 3.1.2 details a scenario where a UAC sends
a CANCEL at the same time that a UAS sends a 200 OK for the
INVITE that the UAC is canceling. When this occurs, it is the
role of the UAC to immediately send a BYE to terminate
the call.
This scenario was reproducible by have a Digium phone with two lines
place a call to a second phone that forwarded the call to the second
line on the original phone. The Digium phone, upon realizing that it
was connecting to itself, would attempt to cancel the call. The timing
of this happened to trigger the aforementioned race condition about
80% of the time. Asterisk was not doing its job of sending a BYE
when receiving a 200 OK on a cancelled INVITE. The result was that
the ast_channel structure was destroyed but the underlying SIP
session, as well as the PJSIP inv_session and dialog, were still
alive. Attempting to perform an action such as a transfer, once in
this state, would result in Asterisk crashing.
The circumstances are now detected properly and the session is ended
as recommended in RFC 5407.
(closes issue AST-1209)
reported by John Bigelow
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The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
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