Add support for SIP over WebSocket.
authorJoshua Colp <jcolp@digium.com>
Mon, 16 Jul 2012 12:35:04 +0000 (12:35 +0000)
committerJoshua Colp <jcolp@digium.com>
Mon, 16 Jul 2012 12:35:04 +0000 (12:35 +0000)
commite9387375700a1b3e78ce44af0495aefe29c6d2a4
treeb46a8344ac5fbb5266e879a8e4409e28a737e327
parentf9c3585d7312e65bda96755fbb8dab21fe54bcd5
Add support for SIP over WebSocket.

This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb.

Review: https://reviewboard.asterisk.org/r/2008

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CHANGES
channels/chan_sip.c
channels/sip/include/sip.h
channels/sip/sdp_crypto.c
channels/sip/security_events.c
configs/sip.conf.sample
include/asterisk/http_websocket.h
res/res_http_websocket.c