application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
New files
- channels/sip/sip.h – A new header for shared #define, enum, and struct
definitions.
- channels/sip/include/sip_utils.h – sip util functions shared among
the all the sip APIs
- channels/sip/include/config_parser.h – sip config-parser API
- channels/sip/config_parser.c – Contains sip.conf parsing helper functions
with unit tests.
- channels/sip/include/reqresp_parser.h – sip request response parser API
- channels/sip/reqresp_parser.c – Contains sip request and response parsing
helper functions with unit tests.
New Unit Tests
- sip_parse_uri_test
- sip_parse_host_test
- sip_parse_register_line_test
Code Refactoring
- All reusable #define, enum, and struct definitions were moved out of chan_sip.c
into sip.h. During this process formatting changes were made to comments
in both sip.h and chan_sip.c in order to better adhere to the coding guidelines.
- The beginnings of three new sip APIs, sip-utils.h, config-parser.h,
reqresp-parser.h using existing chan_sip.c functions.
- parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c
along with unit tests for both functions.
- sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to
config-parser.c along with unit tests for both functions.
Changes to parse_uri()
-removal of the options parameter. It was never used and did not behave correctly.
-additional check for [?header] field. When this field was present, the transport
type was not being set correctly.
----- Overview -----
This patch is introduced with the hope that unit tests for all our sip parsing
functions will be written soon. chan_sip is a huge file, and with the addition of
each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing
we begin refactoring chan_sip, starting with the parsing functions. With each parsing
function we move into a separate helper file, a unit test should accompany it. I've
attempted to lay down the ground work for this change by creating two new parser
helper files (config-parser.c and reqresp-parser.c) and moving all shared structs,
enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything
in Asterisk using unit tests, but string parsing is one area where unit tests make
the most sense. By beginning to restructure the code in this way, chan_sip not only
becomes less bloated, but Asterisk as a whole will become more stable.
Review: https://reviewboard.asterisk.org/r/477/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
