(closes issue #16543) Reported by: crjw Patches: rtp_sin_family.patch uploaded by crjw (license 963) Tested by: crjw, qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1646 lines
56 KiB
C
1646 lines
56 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2008, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Pluggable RTP Architecture
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*
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* \author Joshua Colp <jcolp@digium.com>
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <math.h>
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#include "asterisk/channel.h"
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#include "asterisk/frame.h"
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/manager.h"
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#include "asterisk/options.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/pbx.h"
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#include "asterisk/translate.h"
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/*! Structure that represents an RTP session (instance) */
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struct ast_rtp_instance {
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/*! Engine that is handling this RTP instance */
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struct ast_rtp_engine *engine;
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/*! Data unique to the RTP engine */
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void *data;
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/*! RTP properties that have been set and their value */
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int properties[AST_RTP_PROPERTY_MAX];
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/*! Address that we are expecting RTP to come in to */
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struct sockaddr_in local_address;
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/*! Address that we are sending RTP to */
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struct sockaddr_in remote_address;
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/*! Alternate address that we are receiving RTP from */
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struct sockaddr_in alt_remote_address;
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/*! Instance that we are bridged to if doing remote or local bridging */
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struct ast_rtp_instance *bridged;
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/*! Payload and packetization information */
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struct ast_rtp_codecs codecs;
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/*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
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int timeout;
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/*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
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int holdtimeout;
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/*! DTMF mode in use */
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enum ast_rtp_dtmf_mode dtmf_mode;
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/*! Glue currently in use */
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struct ast_rtp_glue *glue;
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/*! Channel associated with the instance */
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struct ast_channel *chan;
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};
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/*! List of RTP engines that are currently registered */
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static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
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/*! List of RTP glues */
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static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
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/*! The following array defines the MIME Media type (and subtype) for each
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of our codecs, or RTP-specific data type. */
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static const struct ast_rtp_mime_type {
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struct ast_rtp_payload_type payload_type;
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char *type;
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char *subtype;
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unsigned int sample_rate;
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} ast_rtp_mime_types[] = {
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{{1, AST_FORMAT_G723_1}, "audio", "G723", 8000},
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{{1, AST_FORMAT_GSM}, "audio", "GSM", 8000},
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{{1, AST_FORMAT_ULAW}, "audio", "PCMU", 8000},
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{{1, AST_FORMAT_ULAW}, "audio", "G711U", 8000},
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{{1, AST_FORMAT_ALAW}, "audio", "PCMA", 8000},
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{{1, AST_FORMAT_ALAW}, "audio", "G711A", 8000},
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{{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
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{{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
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{{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
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{{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
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{{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
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{{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
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{{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
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{{1, AST_FORMAT_SPEEX}, "audio", "speex", 8000},
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{{1, AST_FORMAT_ILBC}, "audio", "iLBC", 8000},
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/* this is the sample rate listed in the RTP profile for the G.722
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codec, *NOT* the actual sample rate of the media stream
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*/
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{{1, AST_FORMAT_G722}, "audio", "G722", 8000},
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{{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32", 8000},
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{{0, AST_RTP_DTMF}, "audio", "telephone-event", 8000},
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{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event", 8000},
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{{0, AST_RTP_CN}, "audio", "CN", 8000},
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{{1, AST_FORMAT_JPEG}, "video", "JPEG", 90000},
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{{1, AST_FORMAT_PNG}, "video", "PNG", 90000},
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{{1, AST_FORMAT_H261}, "video", "H261", 90000},
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{{1, AST_FORMAT_H263}, "video", "H263", 90000},
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{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998", 90000},
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{{1, AST_FORMAT_H264}, "video", "H264", 90000},
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{{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES", 90000},
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{{1, AST_FORMAT_T140RED}, "text", "RED", 1000},
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{{1, AST_FORMAT_T140}, "text", "T140", 1000},
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{{1, AST_FORMAT_SIREN7}, "audio", "G7221", 16000},
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{{1, AST_FORMAT_SIREN14}, "audio", "G7221", 32000},
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};
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/*!
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* \brief Mapping between Asterisk codecs and rtp payload types
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*
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* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
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* also, our own choices for dynamic payload types. This is our master
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* table for transmission
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*
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* See http://www.iana.org/assignments/rtp-parameters for a list of
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* assigned values
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*/
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static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
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[0] = {1, AST_FORMAT_ULAW},
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#ifdef USE_DEPRECATED_G726
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[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
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#endif
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[3] = {1, AST_FORMAT_GSM},
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[4] = {1, AST_FORMAT_G723_1},
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[5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
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[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
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[7] = {1, AST_FORMAT_LPC10},
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[8] = {1, AST_FORMAT_ALAW},
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[9] = {1, AST_FORMAT_G722},
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[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
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[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
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[13] = {0, AST_RTP_CN},
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[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
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[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
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[18] = {1, AST_FORMAT_G729A},
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[19] = {0, AST_RTP_CN}, /* Also used for CN */
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[26] = {1, AST_FORMAT_JPEG},
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[31] = {1, AST_FORMAT_H261},
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[34] = {1, AST_FORMAT_H263},
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[97] = {1, AST_FORMAT_ILBC},
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[98] = {1, AST_FORMAT_H263_PLUS},
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[99] = {1, AST_FORMAT_H264},
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[101] = {0, AST_RTP_DTMF},
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[102] = {1, AST_FORMAT_SIREN7},
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[103] = {1, AST_FORMAT_H263_PLUS},
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[104] = {1, AST_FORMAT_MP4_VIDEO},
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[105] = {1, AST_FORMAT_T140RED}, /* Real time text chat (with redundancy encoding) */
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[106] = {1, AST_FORMAT_T140}, /* Real time text chat */
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[110] = {1, AST_FORMAT_SPEEX},
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[111] = {1, AST_FORMAT_G726},
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[112] = {1, AST_FORMAT_G726_AAL2},
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[115] = {1, AST_FORMAT_SIREN14},
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[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
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};
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int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
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{
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struct ast_rtp_engine *current_engine;
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/* Perform a sanity check on the engine structure to make sure it has the basics */
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if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
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ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
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return -1;
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}
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/* Link owner module to the RTP engine for reference counting purposes */
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engine->mod = module;
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AST_RWLIST_WRLOCK(&engines);
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/* Ensure that no two modules with the same name are registered at the same time */
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AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
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if (!strcmp(current_engine->name, engine->name)) {
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ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
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AST_RWLIST_UNLOCK(&engines);
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return -1;
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}
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}
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/* The engine survived our critique. Off to the list it goes to be used */
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AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
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AST_RWLIST_UNLOCK(&engines);
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ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
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return 0;
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}
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int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
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{
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struct ast_rtp_engine *current_engine = NULL;
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AST_RWLIST_WRLOCK(&engines);
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if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
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ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
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}
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AST_RWLIST_UNLOCK(&engines);
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return current_engine ? 0 : -1;
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}
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int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
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{
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struct ast_rtp_glue *current_glue = NULL;
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if (ast_strlen_zero(glue->type)) {
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return -1;
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}
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glue->mod = module;
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AST_RWLIST_WRLOCK(&glues);
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AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
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if (!strcasecmp(current_glue->type, glue->type)) {
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ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
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AST_RWLIST_UNLOCK(&glues);
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return -1;
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}
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}
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AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
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AST_RWLIST_UNLOCK(&glues);
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ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
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return 0;
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}
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int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
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{
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struct ast_rtp_glue *current_glue = NULL;
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AST_RWLIST_WRLOCK(&glues);
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if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
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ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
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}
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AST_RWLIST_UNLOCK(&glues);
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return current_glue ? 0 : -1;
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}
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static void instance_destructor(void *obj)
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{
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struct ast_rtp_instance *instance = obj;
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/* Pass us off to the engine to destroy */
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if (instance->data && instance->engine->destroy(instance)) {
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ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
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return;
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}
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/* Drop our engine reference */
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ast_module_unref(instance->engine->mod);
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ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
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}
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int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
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{
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ao2_ref(instance, -1);
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return 0;
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}
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struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sched_context *sched, struct sockaddr_in *sin, void *data)
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{
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struct sockaddr_in address = { 0, };
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struct ast_rtp_instance *instance = NULL;
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struct ast_rtp_engine *engine = NULL;
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AST_RWLIST_RDLOCK(&engines);
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/* If an engine name was specified try to use it or otherwise use the first one registered */
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if (!ast_strlen_zero(engine_name)) {
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AST_RWLIST_TRAVERSE(&engines, engine, entry) {
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if (!strcmp(engine->name, engine_name)) {
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break;
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}
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}
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} else {
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engine = AST_RWLIST_FIRST(&engines);
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}
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/* If no engine was actually found bail out now */
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if (!engine) {
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ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
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AST_RWLIST_UNLOCK(&engines);
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return NULL;
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}
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/* Bump up the reference count before we return so the module can not be unloaded */
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ast_module_ref(engine->mod);
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AST_RWLIST_UNLOCK(&engines);
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/* Allocate a new RTP instance */
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if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
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ast_module_unref(engine->mod);
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return NULL;
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}
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instance->engine = engine;
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instance->local_address.sin_family = AF_INET;
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instance->local_address.sin_addr = sin->sin_addr;
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instance->remote_address.sin_family = AF_INET;
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address.sin_family = AF_INET;
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address.sin_addr = sin->sin_addr;
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ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
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/* And pass it off to the engine to setup */
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if (instance->engine->new(instance, sched, &address, data)) {
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ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
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ao2_ref(instance, -1);
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return NULL;
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}
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ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
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return instance;
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}
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void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
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{
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instance->data = data;
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}
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void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
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{
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return instance->data;
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}
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int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
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{
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return instance->engine->write(instance, frame);
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}
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struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
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{
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return instance->engine->read(instance, rtcp);
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}
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int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
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{
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instance->local_address.sin_addr = address->sin_addr;
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instance->local_address.sin_port = address->sin_port;
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return 0;
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}
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int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
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{
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instance->remote_address.sin_addr = address->sin_addr;
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instance->remote_address.sin_port = address->sin_port;
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/* moo */
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if (instance->engine->remote_address_set) {
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instance->engine->remote_address_set(instance, &instance->remote_address);
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}
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return 0;
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}
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int ast_rtp_instance_set_alt_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
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{
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instance->alt_remote_address.sin_addr = address->sin_addr;
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instance->alt_remote_address.sin_port = address->sin_port;
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/* oink */
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if (instance->engine->alt_remote_address_set) {
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instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
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}
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return 0;
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}
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int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
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{
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if ((address->sin_family != AF_INET) ||
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(address->sin_port != instance->local_address.sin_port) ||
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(address->sin_addr.s_addr != instance->local_address.sin_addr.s_addr)) {
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memcpy(address, &instance->local_address, sizeof(*address));
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return 1;
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}
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return 0;
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}
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int ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
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{
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if ((address->sin_family != AF_INET) ||
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(address->sin_port != instance->remote_address.sin_port) ||
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(address->sin_addr.s_addr != instance->remote_address.sin_addr.s_addr)) {
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memcpy(address, &instance->remote_address, sizeof(*address));
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return 1;
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}
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return 0;
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}
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void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
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{
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if (instance->engine->extended_prop_set) {
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instance->engine->extended_prop_set(instance, property, value);
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}
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}
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void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
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{
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if (instance->engine->extended_prop_get) {
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return instance->engine->extended_prop_get(instance, property);
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}
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return NULL;
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}
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void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
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{
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instance->properties[property] = value;
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if (instance->engine->prop_set) {
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instance->engine->prop_set(instance, property, value);
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}
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}
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int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
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{
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return instance->properties[property];
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}
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struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
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{
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return &instance->codecs;
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}
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|
|
void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
codecs->payloads[i].asterisk_format = 0;
|
|
codecs->payloads[i].code = 0;
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, i, 0, 0);
|
|
}
|
|
}
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (static_RTP_PT[i].code) {
|
|
codecs->payloads[i].asterisk_format = static_RTP_PT[i].asterisk_format;
|
|
codecs->payloads[i].code = static_RTP_PT[i].code;
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (src->payloads[i].code) {
|
|
ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
|
|
dest->payloads[i].asterisk_format = src->payloads[i].asterisk_format;
|
|
dest->payloads[i].code = src->payloads[i].code;
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, i, dest->payloads[i].asterisk_format, dest->payloads[i].code);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
|
|
{
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT || !static_RTP_PT[payload].code) {
|
|
return;
|
|
}
|
|
|
|
codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format;
|
|
codecs->payloads[payload].code = static_RTP_PT[payload].code;
|
|
|
|
ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, codecs->payloads[payload].code);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
|
|
char *mimetype, char *mimesubtype,
|
|
enum ast_rtp_options options,
|
|
unsigned int sample_rate)
|
|
{
|
|
unsigned int i;
|
|
int found = 0;
|
|
|
|
if (pt < 0 || pt >= AST_RTP_MAX_PT)
|
|
return -1; /* bogus payload type */
|
|
|
|
for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
|
|
const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
|
|
|
|
if (strcasecmp(mimesubtype, t->subtype)) {
|
|
continue;
|
|
}
|
|
|
|
if (strcasecmp(mimetype, t->type)) {
|
|
continue;
|
|
}
|
|
|
|
/* if both sample rates have been supplied, and they don't match,
|
|
then this not a match; if one has not been supplied, then the
|
|
rates are not compared */
|
|
if (sample_rate && t->sample_rate &&
|
|
(sample_rate != t->sample_rate)) {
|
|
continue;
|
|
}
|
|
|
|
found = 1;
|
|
codecs->payloads[pt] = t->payload_type;
|
|
|
|
if ((t->payload_type.code == AST_FORMAT_G726) &&
|
|
t->payload_type.asterisk_format &&
|
|
(options & AST_RTP_OPT_G726_NONSTANDARD)) {
|
|
codecs->payloads[pt].code = AST_FORMAT_G726_AAL2;
|
|
}
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
|
|
}
|
|
|
|
break;
|
|
}
|
|
|
|
return (found ? 0 : -2);
|
|
}
|
|
|
|
int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options)
|
|
{
|
|
return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
|
|
{
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
|
|
return;
|
|
}
|
|
|
|
ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
|
|
|
|
codecs->payloads[payload].asterisk_format = 0;
|
|
codecs->payloads[payload].code = 0;
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, payload, 0, 0);
|
|
}
|
|
}
|
|
|
|
struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
|
|
{
|
|
struct ast_rtp_payload_type result = { .asterisk_format = 0, };
|
|
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
|
|
return result;
|
|
}
|
|
|
|
result.asterisk_format = codecs->payloads[payload].asterisk_format;
|
|
result.code = codecs->payloads[payload].code;
|
|
|
|
if (!result.code) {
|
|
result = static_RTP_PT[payload];
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, format_t *astformats, int *nonastformats)
|
|
{
|
|
int i;
|
|
|
|
*astformats = *nonastformats = 0;
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (codecs->payloads[i].code) {
|
|
ast_debug(1, "Incorporating payload %d on %p\n", i, codecs);
|
|
}
|
|
if (codecs->payloads[i].asterisk_format) {
|
|
*astformats |= codecs->payloads[i].code;
|
|
} else {
|
|
*nonastformats |= codecs->payloads[i].code;
|
|
}
|
|
}
|
|
}
|
|
|
|
int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const format_t code)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (codecs->payloads[i].asterisk_format == asterisk_format && codecs->payloads[i].code == code) {
|
|
return i;
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (static_RTP_PT[i].asterisk_format == asterisk_format && static_RTP_PT[i].code == code) {
|
|
return i;
|
|
}
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const format_t code, enum ast_rtp_options options)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); i++) {
|
|
if (ast_rtp_mime_types[i].payload_type.code == code && ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format) {
|
|
if (asterisk_format && (code == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
|
|
return "G726-32";
|
|
} else {
|
|
return ast_rtp_mime_types[i].subtype;
|
|
}
|
|
}
|
|
}
|
|
|
|
return "";
|
|
}
|
|
|
|
unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, format_t code)
|
|
{
|
|
unsigned int i;
|
|
|
|
for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
|
|
if ((ast_rtp_mime_types[i].payload_type.code == code) && (ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format)) {
|
|
return ast_rtp_mime_types[i].sample_rate;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const format_t capability, const int asterisk_format, enum ast_rtp_options options)
|
|
{
|
|
format_t format;
|
|
int found = 0;
|
|
|
|
if (!buf) {
|
|
return NULL;
|
|
}
|
|
|
|
ast_str_append(&buf, 0, "0x%llx (", (unsigned long long) capability);
|
|
|
|
for (format = 1; format < AST_RTP_MAX; format <<= 1) {
|
|
if (capability & format) {
|
|
const char *name = ast_rtp_lookup_mime_subtype2(asterisk_format, format, options);
|
|
ast_str_append(&buf, 0, "%s|", name);
|
|
found = 1;
|
|
}
|
|
}
|
|
|
|
ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
|
|
|
|
return ast_str_buffer(buf);
|
|
}
|
|
|
|
void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
|
|
{
|
|
codecs->pref = *prefs;
|
|
|
|
if (instance && instance->engine->packetization_set) {
|
|
instance->engine->packetization_set(instance, &instance->codecs.pref);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
|
|
{
|
|
return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
|
|
{
|
|
return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
|
|
{
|
|
if (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) {
|
|
return -1;
|
|
}
|
|
|
|
instance->dtmf_mode = dtmf_mode;
|
|
|
|
return 0;
|
|
}
|
|
|
|
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->dtmf_mode;
|
|
}
|
|
|
|
void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
|
|
{
|
|
if (instance->engine->new_source) {
|
|
instance->engine->new_source(instance);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
|
|
{
|
|
return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
|
|
}
|
|
|
|
void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
|
|
{
|
|
if (instance->engine->stop) {
|
|
instance->engine->stop(instance);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
|
|
{
|
|
return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
|
|
}
|
|
|
|
struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
|
|
{
|
|
struct ast_rtp_glue *glue = NULL;
|
|
|
|
AST_RWLIST_RDLOCK(&glues);
|
|
|
|
AST_RWLIST_TRAVERSE(&glues, glue, entry) {
|
|
if (!strcasecmp(glue->type, type)) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
AST_RWLIST_UNLOCK(&glues);
|
|
|
|
return glue;
|
|
}
|
|
|
|
static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
|
|
{
|
|
enum ast_bridge_result res = AST_BRIDGE_FAILED;
|
|
struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
|
|
struct ast_frame *fr = NULL;
|
|
|
|
/* Start locally bridging both instances */
|
|
if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
|
|
ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c0->name, c1->name);
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
return AST_BRIDGE_FAILED_NOWARN;
|
|
}
|
|
if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
|
|
ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c1->name, c0->name);
|
|
if (instance0->engine->local_bridge) {
|
|
instance0->engine->local_bridge(instance0, NULL);
|
|
}
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
return AST_BRIDGE_FAILED_NOWARN;
|
|
}
|
|
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
|
|
instance0->bridged = instance1;
|
|
instance1->bridged = instance0;
|
|
|
|
ast_poll_channel_add(c0, c1);
|
|
|
|
/* Hop into a loop waiting for a frame from either channel */
|
|
cs[0] = c0;
|
|
cs[1] = c1;
|
|
cs[2] = NULL;
|
|
for (;;) {
|
|
/* If the underlying formats have changed force this bridge to break */
|
|
if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) {
|
|
ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
|
|
res = AST_BRIDGE_FAILED_NOWARN;
|
|
break;
|
|
}
|
|
/* Check if anything changed */
|
|
if ((c0->tech_pvt != pvt0) ||
|
|
(c1->tech_pvt != pvt1) ||
|
|
(c0->masq || c0->masqr || c1->masq || c1->masqr) ||
|
|
(c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
|
|
ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
|
|
/* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
|
|
if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) {
|
|
ast_frfree(fr);
|
|
}
|
|
if ((c1->masq || c1->masqr) && (fr = ast_read(c1))) {
|
|
ast_frfree(fr);
|
|
}
|
|
res = AST_BRIDGE_RETRY;
|
|
break;
|
|
}
|
|
/* Wait on a channel to feed us a frame */
|
|
if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
|
|
if (!timeoutms) {
|
|
res = AST_BRIDGE_RETRY;
|
|
break;
|
|
}
|
|
ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
|
|
if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
|
|
break;
|
|
}
|
|
continue;
|
|
}
|
|
/* Read in frame from channel */
|
|
fr = ast_read(who);
|
|
other = (who == c0) ? c1 : c0;
|
|
/* Depending on the frame we may need to break out of our bridge */
|
|
if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
|
|
((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
|
|
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
|
|
/* Record received frame and who */
|
|
*fo = fr;
|
|
*rc = who;
|
|
ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
|
|
res = AST_BRIDGE_COMPLETE;
|
|
break;
|
|
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
|
|
if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
|
|
(fr->subclass.integer == AST_CONTROL_UNHOLD) ||
|
|
(fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
|
|
(fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
|
|
(fr->subclass.integer == AST_CONTROL_T38_PARAMETERS)) {
|
|
/* If we are going on hold, then break callback mode and P2P bridging */
|
|
if (fr->subclass.integer == AST_CONTROL_HOLD) {
|
|
if (instance0->engine->local_bridge) {
|
|
instance0->engine->local_bridge(instance0, NULL);
|
|
}
|
|
if (instance1->engine->local_bridge) {
|
|
instance1->engine->local_bridge(instance1, NULL);
|
|
}
|
|
instance0->bridged = NULL;
|
|
instance1->bridged = NULL;
|
|
} else if (fr->subclass.integer == AST_CONTROL_UNHOLD) {
|
|
if (instance0->engine->local_bridge) {
|
|
instance0->engine->local_bridge(instance0, instance1);
|
|
}
|
|
if (instance1->engine->local_bridge) {
|
|
instance1->engine->local_bridge(instance1, instance0);
|
|
}
|
|
instance0->bridged = instance1;
|
|
instance1->bridged = instance0;
|
|
}
|
|
ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
|
|
ast_frfree(fr);
|
|
} else {
|
|
*fo = fr;
|
|
*rc = who;
|
|
ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, who->name);
|
|
res = AST_BRIDGE_COMPLETE;
|
|
break;
|
|
}
|
|
} else {
|
|
if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
|
|
(fr->frametype == AST_FRAME_DTMF_END) ||
|
|
(fr->frametype == AST_FRAME_VOICE) ||
|
|
(fr->frametype == AST_FRAME_VIDEO) ||
|
|
(fr->frametype == AST_FRAME_IMAGE) ||
|
|
(fr->frametype == AST_FRAME_HTML) ||
|
|
(fr->frametype == AST_FRAME_MODEM) ||
|
|
(fr->frametype == AST_FRAME_TEXT)) {
|
|
ast_write(other, fr);
|
|
}
|
|
|
|
ast_frfree(fr);
|
|
}
|
|
/* Swap priority */
|
|
cs[2] = cs[0];
|
|
cs[0] = cs[1];
|
|
cs[1] = cs[2];
|
|
}
|
|
|
|
/* Stop locally bridging both instances */
|
|
if (instance0->engine->local_bridge) {
|
|
instance0->engine->local_bridge(instance0, NULL);
|
|
}
|
|
if (instance1->engine->local_bridge) {
|
|
instance1->engine->local_bridge(instance1, NULL);
|
|
}
|
|
|
|
instance0->bridged = NULL;
|
|
instance1->bridged = NULL;
|
|
|
|
ast_poll_channel_del(c0, c1);
|
|
|
|
return res;
|
|
}
|
|
|
|
static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1,
|
|
struct ast_rtp_instance *vinstance0, struct ast_rtp_instance *vinstance1, struct ast_rtp_instance *tinstance0,
|
|
struct ast_rtp_instance *tinstance1, struct ast_rtp_glue *glue0, struct ast_rtp_glue *glue1, format_t codec0, format_t codec1, int timeoutms,
|
|
int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
|
|
{
|
|
enum ast_bridge_result res = AST_BRIDGE_FAILED;
|
|
struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
|
|
format_t oldcodec0 = codec0, oldcodec1 = codec1;
|
|
struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,};
|
|
struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,};
|
|
struct ast_frame *fr = NULL;
|
|
|
|
/* Test the first channel */
|
|
if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0))) {
|
|
ast_rtp_instance_get_remote_address(instance1, &ac1);
|
|
if (vinstance1) {
|
|
ast_rtp_instance_get_remote_address(vinstance1, &vac1);
|
|
}
|
|
if (tinstance1) {
|
|
ast_rtp_instance_get_remote_address(tinstance1, &tac1);
|
|
}
|
|
} else {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
|
|
}
|
|
|
|
/* Test the second channel */
|
|
if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0))) {
|
|
ast_rtp_instance_get_remote_address(instance0, &ac0);
|
|
if (vinstance0) {
|
|
ast_rtp_instance_get_remote_address(instance0, &vac0);
|
|
}
|
|
if (tinstance0) {
|
|
ast_rtp_instance_get_remote_address(instance0, &tac0);
|
|
}
|
|
} else {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
|
|
}
|
|
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
|
|
instance0->bridged = instance1;
|
|
instance1->bridged = instance0;
|
|
|
|
ast_poll_channel_add(c0, c1);
|
|
|
|
/* Go into a loop handling any stray frames that may come in */
|
|
cs[0] = c0;
|
|
cs[1] = c1;
|
|
cs[2] = NULL;
|
|
for (;;) {
|
|
/* Check if anything changed */
|
|
if ((c0->tech_pvt != pvt0) ||
|
|
(c1->tech_pvt != pvt1) ||
|
|
(c0->masq || c0->masqr || c1->masq || c1->masqr) ||
|
|
(c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
|
|
ast_debug(1, "Oooh, something is weird, backing out\n");
|
|
res = AST_BRIDGE_RETRY;
|
|
break;
|
|
}
|
|
|
|
/* Check if they have changed their address */
|
|
ast_rtp_instance_get_remote_address(instance1, &t1);
|
|
if (vinstance1) {
|
|
ast_rtp_instance_get_remote_address(vinstance1, &vt1);
|
|
}
|
|
if (tinstance1) {
|
|
ast_rtp_instance_get_remote_address(tinstance1, &tt1);
|
|
}
|
|
if (glue1->get_codec) {
|
|
codec1 = glue1->get_codec(c1);
|
|
}
|
|
|
|
ast_rtp_instance_get_remote_address(instance0, &t0);
|
|
if (vinstance0) {
|
|
ast_rtp_instance_get_remote_address(vinstance0, &vt0);
|
|
}
|
|
if (tinstance0) {
|
|
ast_rtp_instance_get_remote_address(tinstance0, &tt0);
|
|
}
|
|
if (glue0->get_codec) {
|
|
codec0 = glue0->get_codec(c0);
|
|
}
|
|
|
|
if ((inaddrcmp(&t1, &ac1)) ||
|
|
(vinstance1 && inaddrcmp(&vt1, &vac1)) ||
|
|
(tinstance1 && inaddrcmp(&tt1, &tac1)) ||
|
|
(codec1 != oldcodec1)) {
|
|
ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %s)\n",
|
|
c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), ast_getformatname(codec1));
|
|
ast_debug(1, "Oooh, '%s' changed end vaddress to %s:%d (format %s)\n",
|
|
c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), ast_getformatname(codec1));
|
|
ast_debug(1, "Oooh, '%s' changed end taddress to %s:%d (format %s)\n",
|
|
c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), ast_getformatname(codec1));
|
|
ast_debug(1, "Oooh, '%s' was %s:%d/(format %s)\n",
|
|
c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), ast_getformatname(oldcodec1));
|
|
ast_debug(1, "Oooh, '%s' was %s:%d/(format %s)\n",
|
|
c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), ast_getformatname(oldcodec1));
|
|
ast_debug(1, "Oooh, '%s' was %s:%d/(format %s)\n",
|
|
c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), ast_getformatname(oldcodec1));
|
|
if (glue0->update_peer(c0, t1.sin_addr.s_addr ? instance1 : NULL, vt1.sin_addr.s_addr ? vinstance1 : NULL, tt1.sin_addr.s_addr ? tinstance1 : NULL, codec1, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
|
|
}
|
|
memcpy(&ac1, &t1, sizeof(ac1));
|
|
memcpy(&vac1, &vt1, sizeof(vac1));
|
|
memcpy(&tac1, &tt1, sizeof(tac1));
|
|
oldcodec1 = codec1;
|
|
}
|
|
if ((inaddrcmp(&t0, &ac0)) ||
|
|
(vinstance0 && inaddrcmp(&vt0, &vac0)) ||
|
|
(tinstance0 && inaddrcmp(&tt0, &tac0)) ||
|
|
(codec0 != oldcodec0)) {
|
|
ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %s)\n",
|
|
c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), ast_getformatname(codec0));
|
|
ast_debug(1, "Oooh, '%s' was %s:%d/(format %s)\n",
|
|
c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), ast_getformatname(oldcodec0));
|
|
if (glue1->update_peer(c1, t0.sin_addr.s_addr ? instance0 : NULL, vt0.sin_addr.s_addr ? vinstance0 : NULL, tt0.sin_addr.s_addr ? tinstance0 : NULL, codec0, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
|
|
}
|
|
memcpy(&ac0, &t0, sizeof(ac0));
|
|
memcpy(&vac0, &vt0, sizeof(vac0));
|
|
memcpy(&tac0, &tt0, sizeof(tac0));
|
|
oldcodec0 = codec0;
|
|
}
|
|
|
|
/* Wait for frame to come in on the channels */
|
|
if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
|
|
if (!timeoutms) {
|
|
res = AST_BRIDGE_RETRY;
|
|
break;
|
|
}
|
|
ast_debug(1, "Ooh, empty read...\n");
|
|
if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
|
|
break;
|
|
}
|
|
continue;
|
|
}
|
|
fr = ast_read(who);
|
|
other = (who == c0) ? c1 : c0;
|
|
if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
|
|
(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
|
|
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
|
|
/* Break out of bridge */
|
|
*fo = fr;
|
|
*rc = who;
|
|
ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
|
|
res = AST_BRIDGE_COMPLETE;
|
|
break;
|
|
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
|
|
if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
|
|
(fr->subclass.integer == AST_CONTROL_UNHOLD) ||
|
|
(fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
|
|
(fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
|
|
(fr->subclass.integer == AST_CONTROL_T38_PARAMETERS)) {
|
|
if (fr->subclass.integer == AST_CONTROL_HOLD) {
|
|
/* If we someone went on hold we want the other side to reinvite back to us */
|
|
if (who == c0) {
|
|
glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
|
|
} else {
|
|
glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
|
|
}
|
|
} else if (fr->subclass.integer == AST_CONTROL_UNHOLD) {
|
|
/* If they went off hold they should go back to being direct */
|
|
if (who == c0) {
|
|
glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0);
|
|
} else {
|
|
glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0);
|
|
}
|
|
}
|
|
/* Update local address information */
|
|
ast_rtp_instance_get_remote_address(instance0, &t0);
|
|
memcpy(&ac0, &t0, sizeof(ac0));
|
|
ast_rtp_instance_get_remote_address(instance1, &t1);
|
|
memcpy(&ac1, &t1, sizeof(ac1));
|
|
/* Update codec information */
|
|
if (glue0->get_codec && c0->tech_pvt) {
|
|
oldcodec0 = codec0 = glue0->get_codec(c0);
|
|
}
|
|
if (glue1->get_codec && c1->tech_pvt) {
|
|
oldcodec1 = codec1 = glue1->get_codec(c1);
|
|
}
|
|
ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
|
|
ast_frfree(fr);
|
|
} else {
|
|
*fo = fr;
|
|
*rc = who;
|
|
ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, who->name);
|
|
return AST_BRIDGE_COMPLETE;
|
|
}
|
|
} else {
|
|
if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
|
|
(fr->frametype == AST_FRAME_DTMF_END) ||
|
|
(fr->frametype == AST_FRAME_VOICE) ||
|
|
(fr->frametype == AST_FRAME_VIDEO) ||
|
|
(fr->frametype == AST_FRAME_IMAGE) ||
|
|
(fr->frametype == AST_FRAME_HTML) ||
|
|
(fr->frametype == AST_FRAME_MODEM) ||
|
|
(fr->frametype == AST_FRAME_TEXT)) {
|
|
ast_write(other, fr);
|
|
}
|
|
ast_frfree(fr);
|
|
}
|
|
/* Swap priority */
|
|
cs[2] = cs[0];
|
|
cs[0] = cs[1];
|
|
cs[1] = cs[2];
|
|
}
|
|
|
|
if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
|
|
}
|
|
if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
|
|
}
|
|
|
|
instance0->bridged = NULL;
|
|
instance1->bridged = NULL;
|
|
|
|
ast_poll_channel_del(c0, c1);
|
|
|
|
return res;
|
|
}
|
|
|
|
/*!
|
|
* \brief Conditionally unref an rtp instance
|
|
*/
|
|
static void unref_instance_cond(struct ast_rtp_instance **instance)
|
|
{
|
|
if (*instance) {
|
|
ao2_ref(*instance, -1);
|
|
*instance = NULL;
|
|
}
|
|
}
|
|
|
|
enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
|
|
{
|
|
struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
|
|
*vinstance0 = NULL, *vinstance1 = NULL,
|
|
*tinstance0 = NULL, *tinstance1 = NULL;
|
|
struct ast_rtp_glue *glue0, *glue1;
|
|
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
enum ast_bridge_result res = AST_BRIDGE_FAILED;
|
|
format_t codec0 = 0, codec1 = 0;
|
|
int unlock_chans = 1;
|
|
|
|
/* Lock both channels so we can look for the glue that binds them together */
|
|
ast_channel_lock(c0);
|
|
while (ast_channel_trylock(c1)) {
|
|
ast_channel_unlock(c0);
|
|
usleep(1);
|
|
ast_channel_lock(c0);
|
|
}
|
|
|
|
/* Ensure neither channel got hungup during lock avoidance */
|
|
if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
|
|
ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
|
|
goto done;
|
|
}
|
|
|
|
/* Grab glue that binds each channel to something using the RTP engine */
|
|
if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
|
|
ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
|
|
goto done;
|
|
}
|
|
|
|
audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
|
|
video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
|
|
video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
|
|
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
|
|
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
|
|
if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
|
|
res = AST_BRIDGE_FAILED_NOWARN;
|
|
goto done;
|
|
}
|
|
|
|
/* If we need to get DTMF see if we can do it outside of the RTP stream itself */
|
|
if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && instance0->properties[AST_RTP_PROPERTY_DTMF]) {
|
|
res = AST_BRIDGE_FAILED_NOWARN;
|
|
goto done;
|
|
}
|
|
if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && instance1->properties[AST_RTP_PROPERTY_DTMF]) {
|
|
res = AST_BRIDGE_FAILED_NOWARN;
|
|
goto done;
|
|
}
|
|
|
|
/* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */
|
|
if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) {
|
|
res = AST_BRIDGE_FAILED_NOWARN;
|
|
goto done;
|
|
}
|
|
|
|
/* Make sure that codecs match */
|
|
codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0;
|
|
codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0;
|
|
if (codec0 && codec1 && !(codec0 & codec1)) {
|
|
ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n", ast_getformatname(codec0), ast_getformatname(codec1));
|
|
res = AST_BRIDGE_FAILED_NOWARN;
|
|
goto done;
|
|
}
|
|
|
|
instance0->glue = glue0;
|
|
instance1->glue = glue1;
|
|
instance0->chan = c0;
|
|
instance1->chan = c1;
|
|
|
|
/* Depending on the end result for bridging either do a local bridge or remote bridge */
|
|
if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
|
|
ast_verbose(VERBOSE_PREFIX_3 "Locally bridging %s and %s\n", c0->name, c1->name);
|
|
res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt);
|
|
} else {
|
|
ast_verbose(VERBOSE_PREFIX_3 "Remotely bridging %s and %s\n", c0->name, c1->name);
|
|
res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
|
|
tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags,
|
|
fo, rc, c0->tech_pvt, c1->tech_pvt);
|
|
}
|
|
|
|
instance0->glue = NULL;
|
|
instance1->glue = NULL;
|
|
instance0->chan = NULL;
|
|
instance1->chan = NULL;
|
|
|
|
unlock_chans = 0;
|
|
|
|
done:
|
|
if (unlock_chans) {
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
}
|
|
|
|
unref_instance_cond(&instance0);
|
|
unref_instance_cond(&instance1);
|
|
unref_instance_cond(&vinstance0);
|
|
unref_instance_cond(&vinstance1);
|
|
unref_instance_cond(&tinstance0);
|
|
unref_instance_cond(&tinstance1);
|
|
|
|
return res;
|
|
}
|
|
|
|
struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->bridged;
|
|
}
|
|
|
|
void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
|
|
{
|
|
struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
|
|
*vinstance0 = NULL, *vinstance1 = NULL,
|
|
*tinstance0 = NULL, *tinstance1 = NULL;
|
|
struct ast_rtp_glue *glue0, *glue1;
|
|
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
format_t codec0 = 0, codec1 = 0;
|
|
int res = 0;
|
|
|
|
/* Lock both channels so we can look for the glue that binds them together */
|
|
ast_channel_lock(c0);
|
|
while (ast_channel_trylock(c1)) {
|
|
ast_channel_unlock(c0);
|
|
usleep(1);
|
|
ast_channel_lock(c0);
|
|
}
|
|
|
|
/* Grab glue that binds each channel to something using the RTP engine */
|
|
if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
|
|
ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
|
|
goto done;
|
|
}
|
|
|
|
audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
|
|
video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
|
|
video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
|
|
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
|
|
codec0 = glue0->get_codec(c0);
|
|
}
|
|
if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
|
|
codec1 = glue1->get_codec(c1);
|
|
}
|
|
|
|
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
|
|
if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
|
|
goto done;
|
|
}
|
|
|
|
/* Make sure we have matching codecs */
|
|
if (!(codec0 & codec1)) {
|
|
goto done;
|
|
}
|
|
|
|
ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
|
|
|
|
if (vinstance0 && vinstance1) {
|
|
ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
|
|
}
|
|
if (tinstance0 && tinstance1) {
|
|
ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
|
|
}
|
|
|
|
res = 0;
|
|
|
|
done:
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
|
|
unref_instance_cond(&instance0);
|
|
unref_instance_cond(&instance1);
|
|
unref_instance_cond(&vinstance0);
|
|
unref_instance_cond(&vinstance1);
|
|
unref_instance_cond(&tinstance0);
|
|
unref_instance_cond(&tinstance1);
|
|
|
|
if (!res) {
|
|
ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
|
|
{
|
|
struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
|
|
*vinstance0 = NULL, *vinstance1 = NULL,
|
|
*tinstance0 = NULL, *tinstance1 = NULL;
|
|
struct ast_rtp_glue *glue0, *glue1;
|
|
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
format_t codec0 = 0, codec1 = 0;
|
|
int res = 0;
|
|
|
|
/* If there is no second channel just immediately bail out, we are of no use in that scenario */
|
|
if (!c1) {
|
|
return -1;
|
|
}
|
|
|
|
/* Lock both channels so we can look for the glue that binds them together */
|
|
ast_channel_lock(c0);
|
|
while (ast_channel_trylock(c1)) {
|
|
ast_channel_unlock(c0);
|
|
usleep(1);
|
|
ast_channel_lock(c0);
|
|
}
|
|
|
|
/* Grab glue that binds each channel to something using the RTP engine */
|
|
if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
|
|
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
|
|
goto done;
|
|
}
|
|
|
|
audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
|
|
video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
|
|
video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
|
|
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
|
|
codec0 = glue0->get_codec(c0);
|
|
}
|
|
if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
|
|
codec1 = glue1->get_codec(c1);
|
|
}
|
|
|
|
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
|
|
if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
|
|
goto done;
|
|
}
|
|
|
|
/* Make sure we have matching codecs */
|
|
if (!(codec0 & codec1)) {
|
|
goto done;
|
|
}
|
|
|
|
/* Bridge media early */
|
|
if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
|
|
}
|
|
|
|
res = 0;
|
|
|
|
done:
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
|
|
unref_instance_cond(&instance0);
|
|
unref_instance_cond(&instance1);
|
|
unref_instance_cond(&vinstance0);
|
|
unref_instance_cond(&vinstance1);
|
|
unref_instance_cond(&tinstance0);
|
|
unref_instance_cond(&tinstance1);
|
|
|
|
if (!res) {
|
|
ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
|
|
{
|
|
return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
|
|
}
|
|
|
|
int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
|
|
{
|
|
return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
|
|
{
|
|
return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
|
|
}
|
|
|
|
char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
|
|
{
|
|
struct ast_rtp_instance_stats stats = { 0, };
|
|
enum ast_rtp_instance_stat stat;
|
|
|
|
/* Determine what statistics we will need to retrieve based on field passed in */
|
|
if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
|
|
stat = AST_RTP_INSTANCE_STAT_ALL;
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
|
|
stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
|
|
stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
|
|
stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
|
|
} else {
|
|
return NULL;
|
|
}
|
|
|
|
/* Attempt to actually retrieve the statistics we need to generate the quality string */
|
|
if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
|
|
return NULL;
|
|
}
|
|
|
|
/* Now actually fill the buffer with the good information */
|
|
if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
|
|
snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%u;rxcount=%u;txjitter=%u;txcount=%u;rlp=%u;rtt=%u",
|
|
stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
|
|
snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
|
|
stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter));
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
|
|
snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
|
|
stats.local_minrxploss, stats.local_maxrxploss, stats.local_normdevrxploss, sqrt(stats.local_stdevrxploss), stats.remote_minrxploss, stats.remote_maxrxploss, stats.remote_normdevrxploss, sqrt(stats.remote_stdevrxploss));
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
|
|
snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
|
|
}
|
|
|
|
return buf;
|
|
}
|
|
|
|
void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
|
|
{
|
|
char quality_buf[AST_MAX_USER_FIELD], *quality;
|
|
struct ast_channel *bridge = ast_bridged_channel(chan);
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
|
|
}
|
|
}
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
|
|
}
|
|
}
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
|
|
}
|
|
}
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
|
|
}
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, format_t format)
|
|
{
|
|
return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, format_t format)
|
|
{
|
|
return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
|
|
{
|
|
struct ast_rtp_glue *glue;
|
|
struct ast_rtp_instance *peer_instance = NULL;
|
|
int res = -1;
|
|
|
|
if (!instance->engine->make_compatible) {
|
|
return -1;
|
|
}
|
|
|
|
ast_channel_lock(peer);
|
|
|
|
if (!(glue = ast_rtp_instance_get_glue(peer->tech->type))) {
|
|
ast_channel_unlock(peer);
|
|
return -1;
|
|
}
|
|
|
|
glue->get_rtp_info(peer, &peer_instance);
|
|
|
|
if (!peer_instance || peer_instance->engine != instance->engine) {
|
|
ast_channel_unlock(peer);
|
|
ao2_ref(peer_instance, -1);
|
|
peer_instance = NULL;
|
|
return -1;
|
|
}
|
|
|
|
res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
|
|
|
|
ast_channel_unlock(peer);
|
|
|
|
ao2_ref(peer_instance, -1);
|
|
peer_instance = NULL;
|
|
|
|
return res;
|
|
}
|
|
|
|
format_t ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, format_t to_endpoint, format_t to_asterisk)
|
|
{
|
|
format_t formats;
|
|
|
|
if (instance->engine->available_formats && (formats = instance->engine->available_formats(instance, to_endpoint, to_asterisk))) {
|
|
return formats;
|
|
}
|
|
|
|
return ast_translate_available_formats(to_endpoint, to_asterisk);
|
|
}
|
|
|
|
int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine->activate ? instance->engine->activate(instance) : 0;
|
|
}
|
|
|
|
void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username)
|
|
{
|
|
if (instance->engine->stun_request) {
|
|
instance->engine->stun_request(instance, suggestion, username);
|
|
}
|
|
}
|
|
|
|
void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
|
|
{
|
|
instance->timeout = timeout;
|
|
}
|
|
|
|
void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
|
|
{
|
|
instance->holdtimeout = timeout;
|
|
}
|
|
|
|
int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->timeout;
|
|
}
|
|
|
|
int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->holdtimeout;
|
|
}
|
|
|
|
struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine;
|
|
}
|
|
|
|
struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->glue;
|
|
}
|
|
|
|
struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->chan;
|
|
}
|