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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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9
channels/sip/include/sip.h
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@@ -35,6 +35,7 @@
#include "asterisk/indications.h"
#include "asterisk/security_events.h"
#include "asterisk/features.h"
#include "asterisk/http_websocket.h"
#ifndef FALSE
#define FALSE 0
@@ -369,10 +370,11 @@
#define SIP_PAGE3_NAT_AUTO_RPORT (1 << 2) /*!< DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected */
#define SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3) /*!< DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected */
#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 4) /*!< DP: Only send direct media reinvites on outgoing calls */
#define SIP_PAGE3_USE_AVPF (1 << 5) /*!< DGP: Support a minimal AVPF-compatible profile */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF)
#define CHECK_AUTH_BUF_INITLEN 256
@@ -564,6 +566,8 @@ enum sip_transport {
SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
SIP_TRANSPORT_WS = 1 << 3, /*!< WebSocket, unsecure */
SIP_TRANSPORT_WSS = 1 << 4, /*!< WebSocket, secure */
};
/*! \brief Automatic peer registration behavior
@@ -769,6 +773,7 @@ struct sip_socket {
int fd; /*!< Filed descriptor, the actual socket */
uint16_t port;
struct ast_tcptls_session_instance *tcptls_session; /* If tcp or tls, a socket manager */
struct ast_websocket *ws_session; /*! If ws or wss, a WebSocket session */
};
/*! \brief sip_request: The data grabbed from the UDP socket
@@ -1284,7 +1289,7 @@ struct sip_peer {
enum sip_transport default_outbound_transport; /*!< Peer Registration may change the default outbound transport.
If register expires, default should be reset. to this value */
/* things that don't belong in flags */
unsigned short transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
unsigned short transports:5; /*!< Transports (enum sip_transport) that are acceptable for this peer */
unsigned short is_realtime:1; /*!< this is a 'realtime' peer */
unsigned short rt_fromcontact:1;/*!< copy fromcontact from realtime */
unsigned short host_dynamic:1; /*!< Dynamic Peers register with Asterisk */