Doxygen updates
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48277 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -81,8 +81,39 @@
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\arg translate.h - Transcoding support functions
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\arg \ref channel_drivers - Implemented channel drivers
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\arg \ref Def_Frame Asterisk Multimedia Frames
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\arg \ref Def_Bridge
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*/
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/*! \page Def_Bridge Asterisk Channel Bridges
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In Asterisk, there's several media bridges.
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The Core bridge handles two channels (a "phone call") and bridge
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them together.
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The conference bridge (meetme) handles several channels simultaneosly
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with the support of an external timer (zaptel timer). This is used
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not only by the Conference application (meetme) but also by the
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page application and the SLA system introduced in 1.4.
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The conference bridge does not handle video.
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When two channels of the same type connect, the channel driver
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or the media subsystem used by the channel driver (i.e. RTP)
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can create a native bridge without sending media through the
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core.
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Native briding can be disabled by a number of reasons,
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like DTMF being needed by the core or codecs being incompatible
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so a transcoding module is needed.
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References:
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- ast_channel_early_bridge()
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- ast_channel_bridge()
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- app_meetme.c
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- \ref AstRTPbridge
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- ast_rtp_bridge()
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- \ref Def_Channel
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*/
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/*! \page AstFileDesc File descriptors
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Asterisk File descriptors are connected to each channel (see \ref Def_Channel)
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@ -48,8 +48,11 @@ extern "C" {
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/*! Maximum RTP-specific code */
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#define AST_RTP_MAX AST_RTP_CISCO_DTMF
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/*! Maxmum number of payload defintions for a RTP session */
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#define MAX_RTP_PT 256
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#define FLAG_3389_WARNING (1 << 0)
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enum ast_rtp_options {
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AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
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};
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@ -65,6 +68,8 @@ enum ast_rtp_get_result {
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struct ast_rtp;
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/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem
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*/
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struct ast_rtp_protocol {
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/*! Get RTP struct, or NULL if unwilling to transfer */
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enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
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@ -78,8 +83,7 @@ struct ast_rtp_protocol {
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};
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#define FLAG_3389_WARNING (1 << 0)
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/*! RTP callback structure */
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typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
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/*!
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@ -122,11 +126,13 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
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struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
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/*! Destroy RTP session */
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void ast_rtp_destroy(struct ast_rtp *rtp);
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void ast_rtp_reset(struct ast_rtp *rtp);
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void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
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/*! Stop RTP session, do not destroy structure */
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void ast_rtp_stop(struct ast_rtp *rtp);
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void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
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@ -189,10 +195,18 @@ void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
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/*! \brief Enable STUN capability */
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void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
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/*! \brief Send STUN request (??) */
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void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
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/*! \brief The RTP bridge.
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\arg \ref AstRTPbridge
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*/
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int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
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/*! \brief Register an RTP channel client */
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int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
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/*! \brief Unregister an RTP channel client */
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void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
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@ -201,22 +215,22 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
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having to send a re-invite later */
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int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
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void ast_rtp_stop(struct ast_rtp *rtp);
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/*! \brief Return RTCP quality string */
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char *ast_rtp_get_quality(struct ast_rtp *rtp);
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/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
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int ast_rtcp_send_h261fur(void *data);
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void ast_rtp_init(void);
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int ast_rtp_reload(void);
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char *ast_rtp_get_quality(struct ast_rtp *rtp); /*! \brief Return RTCP quality string */
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void ast_rtp_init(void); /*! Initialize RTP subsystem */
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int ast_rtp_reload(void); /*! reload rtp configuration */
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/*! Set codec preference */
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int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
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/*! Get codec preference */
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struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
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/*! get format from predefined dynamic payload format */
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int ast_rtp_codec_getformat(int pt);
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/*! \brief Set rtp timeout */
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29
main/rtp.c
29
main/rtp.c
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@ -3142,7 +3142,34 @@ static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast
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/*! \brief Bridge calls. If possible and allowed, initiate
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re-invite so the peers exchange media directly outside
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of Asterisk. */
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of Asterisk.
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*/
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/*! \page AstRTPbridge The Asterisk RTP bridge
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The RTP bridge is called from the channel drivers that are using the RTP
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subsystem in Asterisk - like SIP, H.323 and Jingle/Google Talk.
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This bridge aims to offload the Asterisk server by setting up
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the media stream directly between the endpoints, keeping the
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signalling in Asterisk.
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It checks with the channel driver, using a callback function, if
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there are possibilities for a remote bridge.
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If this fails, the bridge hands off to the core bridge. Reasons
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can be NAT support needed, DTMF features in audio needed by
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the PBX for transfers or spying/monitoring on channels.
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If transcoding is needed - we can't do a remote bridge.
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If only NAT support is needed, we're using Asterisk in
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RTP proxy mode with the p2p RTP bridge, basically
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forwarding incoming audio packets to the outbound
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stream on a network level.
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References:
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- ast_rtp_bridge()
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- ast_channel_early_bridge()
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- ast_channel_bridge()
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*/
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enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
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{
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struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
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