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    365ae752
    res_http_websocket: Close websocket correctly and use careful fwrite · 365ae752
    Matthew Jordan authored
    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
    365ae752
    History
    res_http_websocket: Close websocket correctly and use careful fwrite
    Matthew Jordan authored
    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