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Add support for SIP over WebSocket.

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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
This commit is contained in:
Joshua Colp
2012-07-16 12:35:04 +00:00
parent f9c3585d73
commit e938737570
8 changed files with 263 additions and 139 deletions
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2
channels/sip/security_events.c
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@@ -45,8 +45,10 @@ static enum ast_security_event_transport_type security_event_get_transport(const
case SIP_TRANSPORT_UDP:
return AST_SECURITY_EVENT_TRANSPORT_UDP;
case SIP_TRANSPORT_TCP:
case SIP_TRANSPORT_WS:
return AST_SECURITY_EVENT_TRANSPORT_TCP;
case SIP_TRANSPORT_TLS:
case SIP_TRANSPORT_WSS:
return AST_SECURITY_EVENT_TRANSPORT_TLS;
}