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e9387375

Commit e9387375 authored 12 years ago by Joshua Colp

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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

parent f9c3585d

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Changes 8

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CHANGES 2 additions, 0 deletions

CHANGES
channels/chan_sip.c 224 additions, 36 deletions

channels/chan_sip.c
channels/sip/include/sip.h 7 additions, 2 deletions

channels/sip/include/sip.h
channels/sip/sdp_crypto.c 1 addition, 1 deletion

channels/sip/sdp_crypto.c
channels/sip/security_events.c 2 additions, 0 deletions

channels/sip/security_events.c
configs/sip.conf.sample 5 additions, 2 deletions

configs/sip.conf.sample
include/asterisk/http_websocket.h 5 additions, 98 deletions

include/asterisk/http_websocket.h
res/res_http_websocket.c 17 additions, 0 deletions

res/res_http_websocket.c

with 263 additions and 139 deletions

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