res_http_websocket: Close websocket correctly and use careful fwrite
authorMatthew Jordan <mjordan@digium.com>
Thu, 26 Jun 2014 12:21:14 +0000 (12:21 +0000)
committerMatthew Jordan <mjordan@digium.com>
Thu, 26 Jun 2014 12:21:14 +0000 (12:21 +0000)
commit365ae7523b45f18abb1418f498561cc2c8cbf680
tree2d1ce4e889fedf5885299baef55a16df464f7a21
parentd171e0b2e96ca1cc2cf6c53cdd9d5a3c876be91b
res_http_websocket: Close websocket correctly and use careful fwrite

When a client takes a long time to process information received from Asterisk,
a write operation using fwrite may fail to write all information. This causes
the underlying file stream to be in an unknown state, such that the socket
must be disconnected. Unfortunately, there are two problems with this in
Asterisk's existing websocket code:
1. Periodically, during the read loop, Asterisk must write to the connected
   websocket to respond to pings. As such, Asterisk maintains a reference to
   the session during the loop. When ast_http_websocket_write fails, it may
   cause the session to decrement its ref count, but this in and of itself
   does not break the read loop. The read loop's write, on the other hand,
   does not break the loop if it fails. This causes the socket to get in a
   'stuck' state, preventing the client from reconnecting to the server.
2. More importantly, however, is that the fwrite in ast_http_websocket_write
   fails with a large volume of data when the client takes awhile to process
   the information. When it does fail, it fails writing only a portion of
   the bytes. With some debugging, it was shown that this was failing in a
   similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite
   with a long enough timeout solved the problem.

Note that this version of the patch, unlike r417310 in Asterisk 11, exposes
configuration options beyond just chan_sip's sip.conf. Configuration options
to configure the write timeout have also been added to pjsip.conf and ari.conf.

#ASTERISK-23917 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3624/
........

Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
16 files changed:
UPGRADE.txt
channels/chan_sip.c
channels/sip/include/sip.h
configs/ari.conf.sample
configs/pjsip.conf.sample
configs/sip.conf.sample
include/asterisk/http_websocket.h
include/asterisk/res_pjsip.h
res/ari/ari_websockets.c
res/ari/config.c
res/ari/internal.h
res/res_ari.c
res/res_http_websocket.c
res/res_pjsip.c
res/res_pjsip/config_transport.c
res/res_pjsip_transport_websocket.c