* Fixed an issue in webchan_write() where we weren't detecting equivalent
codecs properly.
* Added the "p" dialstring option that puts the channel driver in
"passthrough" mode where it will not attempt to re-frame or re-time
media coming in over the websocket from the remote app. This can be used
for any codec but MUST be used for codecs that use packet headers or whose
data stream can't be broken up on arbitrary byte boundaries. In this case,
the remote app is fully responsible for correctly framing and timing media
sent to Asterisk and the MEDIA text commands that could be sent over the
websocket are disabled. Currently, passthrough mode is automatically set
for the opus, speex and g729 codecs.
* Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
ensure proper translation paths are set up when switching between native
frames and slin silence frames. This fixes an issue with codec errors
when transcode_via_sln=yes.
Resolves: #1462
* Added a new option to the WebSocket dial string to capture the additional
URI parameters.
* Added a new API ast_uri_verify_encoded() that verifies that a string
either doesn't need URI encoding or that it has already been encoded.
* Added a new API ast_websocket_client_add_uri_params() to add the params
to the client websocket session.
* Added XML documentation that will show up with `core show application Dial`
that shows how to use it.
Resolves: #1352
UserNote: A new WebSocket channel driver option `v` has been added to the
Dial application that allows you to specify additional URI parameters on
outgoing connections. Run `core show application Dial` from the Asterisk CLI
to see how to use it.
ast_websocket_read() receives data into a fixed 64K buffer then continually
reallocates a final buffer that, after all continuation frames have been
received, is the exact length of the data received and returns that to the
caller. process_text_message() in chan_websocket was attempting to set a
NULL terminator on the received payload assuming the payload buffer it
received was the large 64K buffer. The assumption was incorrect so when it
tried to set a NULL terminator on the payload, it could, depending on the
state of the heap at the time, cause heap corruption.
process_text_message() now allocates its own payload_len + 1 sized buffer,
copies the payload received from ast_websocket_read() into it then NULL
terminates it prevent the possibility of the overrun and corruption.
Resolves: #1384
In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from
the queue but does not reset the frame_queue_length counter.
As a result, the driver incorrectly thinks the queue is full after flushing,
which prevents new multimedia frames from being sent, especially after multiple
flush commands.
This fix sets frame_queue_length to 0 after flushing, ensuring the queue state
is consistent with its actual content.
Fixes: #1304
* Created chan_websocket which can exchange media over both inbound and
outbound websockets which the driver will frame and time.
See http://s.asterisk.net/mow for more information.
* res_http_websocket: Made defines for max message size public and converted
a few nuisance verbose messages to debugs.
* main/channel.c: Changed an obsolete nuisance error to a debug.
* ARI channels: Updated externalMedia to include chan_websocket as a supported
transport.
UserNote: A new channel driver "chan_websocket" is now available. It can
exchange media over both inbound and outbound websockets and will both frame
and re-time the media it receives.
See http://s.asterisk.net/mow for more information.
UserNote: The ARI channels/externalMedia API now includes support for the
WebSocket transport provided by chan_websocket.
