/* $Id$ */ /* * Copyright (C) 2008-2009 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 Benny Prijono * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include #include #define THIS_FILE "pjsua_media.c" #define DEFAULT_RTP_PORT 4000 #define NULL_SND_DEV_ID -99 #ifndef PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT # define PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT 0 #endif /* Next RTP port to be used */ static pj_uint16_t next_rtp_port; /* Close existing sound device */ static void close_snd_dev(void); static void pjsua_media_config_dup(pj_pool_t *pool, pjsua_media_config *dst, const pjsua_media_config *src) { pj_memcpy(dst, src, sizeof(*src)); pj_strdup(pool, &dst->turn_server, &src->turn_server); pj_stun_auth_cred_dup(pool, &dst->turn_auth_cred, &src->turn_auth_cred); } /** * Init media subsystems. */ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) { pj_str_t codec_id = {NULL, 0}; unsigned opt; pj_status_t status; /* To suppress warning about unused var when all codecs are disabled */ PJ_UNUSED_ARG(codec_id); /* Copy configuration */ pjsua_media_config_dup(pjsua_var.pool, &pjsua_var.media_cfg, cfg); /* Normalize configuration */ if (pjsua_var.media_cfg.snd_clock_rate == 0) { pjsua_var.media_cfg.snd_clock_rate = pjsua_var.media_cfg.clock_rate; } if (pjsua_var.media_cfg.has_ioqueue && pjsua_var.media_cfg.thread_cnt == 0) { pjsua_var.media_cfg.thread_cnt = 1; } if (pjsua_var.media_cfg.max_media_ports < pjsua_var.ua_cfg.max_calls) { pjsua_var.media_cfg.max_media_ports = pjsua_var.ua_cfg.max_calls + 2; } /* Create media endpoint. */ status = pjmedia_endpt_create(&pjsua_var.cp.factory, pjsua_var.media_cfg.has_ioqueue? NULL : pjsip_endpt_get_ioqueue(pjsua_var.endpt), pjsua_var.media_cfg.thread_cnt, &pjsua_var.med_endpt); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Media stack initialization has returned error", status); return status; } /* Register all codecs */ #if PJMEDIA_HAS_SPEEX_CODEC /* Register speex. */ status = pjmedia_codec_speex_init(pjsua_var.med_endpt, 0, pjsua_var.media_cfg.quality, -1); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing Speex codec", status); return status; } /* Set speex/16000 to higher priority*/ codec_id = pj_str("speex/16000"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_NORMAL+2); /* Set speex/8000 to next higher priority*/ codec_id = pj_str("speex/8000"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_NORMAL+1); #endif /* PJMEDIA_HAS_SPEEX_CODEC */ #if PJMEDIA_HAS_ILBC_CODEC /* Register iLBC. */ status = pjmedia_codec_ilbc_init( pjsua_var.med_endpt, pjsua_var.media_cfg.ilbc_mode); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing iLBC codec", status); return status; } #endif /* PJMEDIA_HAS_ILBC_CODEC */ #if PJMEDIA_HAS_GSM_CODEC /* Register GSM */ status = pjmedia_codec_gsm_init(pjsua_var.med_endpt); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing GSM codec", status); return status; } #endif /* PJMEDIA_HAS_GSM_CODEC */ #if PJMEDIA_HAS_G711_CODEC /* Register PCMA and PCMU */ status = pjmedia_codec_g711_init(pjsua_var.med_endpt); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing G711 codec", status); return status; } #endif /* PJMEDIA_HAS_G711_CODEC */ #if PJMEDIA_HAS_G722_CODEC status = pjmedia_codec_g722_init( pjsua_var.med_endpt ); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing G722 codec", status); return status; } #endif /* PJMEDIA_HAS_G722_CODEC */ #if PJMEDIA_HAS_INTEL_IPP /* Register IPP codecs */ status = pjmedia_codec_ipp_init(pjsua_var.med_endpt); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing IPP codecs", status); return status; } #endif /* PJMEDIA_HAS_INTEL_IPP */ #if PJMEDIA_HAS_L16_CODEC /* Register L16 family codecs, but disable all */ status = pjmedia_codec_l16_init(pjsua_var.med_endpt, 0); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing L16 codecs", status); return status; } /* Disable ALL L16 codecs */ codec_id = pj_str("L16"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_DISABLED); #endif /* PJMEDIA_HAS_L16_CODEC */ /* Save additional conference bridge parameters for future * reference. */ pjsua_var.mconf_cfg.channel_count = pjsua_var.media_cfg.channel_count; pjsua_var.mconf_cfg.bits_per_sample = 16; pjsua_var.mconf_cfg.samples_per_frame = pjsua_var.media_cfg.clock_rate * pjsua_var.mconf_cfg.channel_count * pjsua_var.media_cfg.audio_frame_ptime / 1000; /* Init options for conference bridge. */ opt = PJMEDIA_CONF_NO_DEVICE; if (pjsua_var.media_cfg.quality >= 3 && pjsua_var.media_cfg.quality <= 4) { opt |= PJMEDIA_CONF_SMALL_FILTER; } else if (pjsua_var.media_cfg.quality < 3) { opt |= PJMEDIA_CONF_USE_LINEAR; } /* Init conference bridge. */ status = pjmedia_conf_create(pjsua_var.pool, pjsua_var.media_cfg.max_media_ports, pjsua_var.media_cfg.clock_rate, pjsua_var.mconf_cfg.channel_count, pjsua_var.mconf_cfg.samples_per_frame, pjsua_var.mconf_cfg.bits_per_sample, opt, &pjsua_var.mconf); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error creating conference bridge", status); return status; } /* Create null port just in case user wants to use null sound. */ status = pjmedia_null_port_create(pjsua_var.pool, pjsua_var.media_cfg.clock_rate, pjsua_var.mconf_cfg.channel_count, pjsua_var.mconf_cfg.samples_per_frame, pjsua_var.mconf_cfg.bits_per_sample, &pjsua_var.null_port); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); /* Perform NAT detection */ pjsua_detect_nat_type(); return PJ_SUCCESS; } /* * Create RTP and RTCP socket pair, and possibly resolve their public * address via STUN. */ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, pjmedia_sock_info *skinfo) { enum { RTP_RETRY = 100 }; int i; pj_sockaddr_in bound_addr; pj_sockaddr_in mapped_addr[2]; pj_status_t status = PJ_SUCCESS; char addr_buf[PJ_INET6_ADDRSTRLEN+2]; pj_sock_t sock[2]; /* Make sure STUN server resolution has completed */ status = pjsua_resolve_stun_server(PJ_TRUE); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error resolving STUN server", status); return status; } if (next_rtp_port == 0) next_rtp_port = (pj_uint16_t)cfg->port; for (i=0; i<2; ++i) sock[i] = PJ_INVALID_SOCKET; bound_addr.sin_addr.s_addr = PJ_INADDR_ANY; if (cfg->bound_addr.slen) { status = pj_sockaddr_in_set_str_addr(&bound_addr, &cfg->bound_addr); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to resolve transport bind address", status); return status; } } /* Loop retry to bind RTP and RTCP sockets. */ for (i=0; ipublic_addr.slen) { status = pj_sockaddr_in_init(&mapped_addr[0], &cfg->public_addr, (pj_uint16_t)next_rtp_port); if (status != PJ_SUCCESS) goto on_error; status = pj_sockaddr_in_init(&mapped_addr[1], &cfg->public_addr, (pj_uint16_t)(next_rtp_port+1)); if (status != PJ_SUCCESS) goto on_error; break; } else { if (bound_addr.sin_addr.s_addr == 0) { pj_sockaddr addr; /* Get local IP address. */ status = pj_gethostip(pj_AF_INET(), &addr); if (status != PJ_SUCCESS) goto on_error; bound_addr.sin_addr.s_addr = addr.ipv4.sin_addr.s_addr; } for (i=0; i<2; ++i) { pj_sockaddr_in_init(&mapped_addr[i], NULL, 0); mapped_addr[i].sin_addr.s_addr = bound_addr.sin_addr.s_addr; } mapped_addr[0].sin_port=pj_htons((pj_uint16_t)next_rtp_port); mapped_addr[1].sin_port=pj_htons((pj_uint16_t)(next_rtp_port+1)); break; } } if (sock[0] == PJ_INVALID_SOCKET) { PJ_LOG(1,(THIS_FILE, "Unable to find appropriate RTP/RTCP ports combination")); goto on_error; } skinfo->rtp_sock = sock[0]; pj_memcpy(&skinfo->rtp_addr_name, &mapped_addr[0], sizeof(pj_sockaddr_in)); skinfo->rtcp_sock = sock[1]; pj_memcpy(&skinfo->rtcp_addr_name, &mapped_addr[1], sizeof(pj_sockaddr_in)); PJ_LOG(4,(THIS_FILE, "RTP socket reachable at %s", pj_sockaddr_print(&skinfo->rtp_addr_name, addr_buf, sizeof(addr_buf), 3))); PJ_LOG(4,(THIS_FILE, "RTCP socket reachable at %s", pj_sockaddr_print(&skinfo->rtcp_addr_name, addr_buf, sizeof(addr_buf), 3))); next_rtp_port += 2; return PJ_SUCCESS; on_error: for (i=0; i<2; ++i) { if (sock[i] != PJ_INVALID_SOCKET) pj_sock_close(sock[i]); } return status; } /* Check if sound device is idle. */ static void check_snd_dev_idle() { /* Activate sound device auto-close timer if sound device is idle. * It is idle when there is no port connection in the bridge. */ if ((pjsua_var.snd_port!=NULL || pjsua_var.null_snd!=NULL) && pjsua_var.snd_idle_timer.id == PJ_FALSE && pjmedia_conf_get_connect_count(pjsua_var.mconf) == 0 && pjsua_var.media_cfg.snd_auto_close_time >= 0) { pj_time_val delay; delay.msec = 0; delay.sec = pjsua_var.media_cfg.snd_auto_close_time; pjsua_var.snd_idle_timer.id = PJ_TRUE; pjsip_endpt_schedule_timer(pjsua_var.endpt, &pjsua_var.snd_idle_timer, &delay); } } /* Timer callback to close sound device */ static void close_snd_timer_cb( pj_timer_heap_t *th, pj_timer_entry *entry) { PJ_UNUSED_ARG(th); PJSUA_LOCK(); if (entry->id) { PJ_LOG(4,(THIS_FILE,"Closing sound device after idle for %d seconds", pjsua_var.media_cfg.snd_auto_close_time)); entry->id = PJ_FALSE; close_snd_dev(); } PJSUA_UNLOCK(); } /* * Start pjsua media subsystem. */ pj_status_t pjsua_media_subsys_start(void) { pj_status_t status; /* Create media for calls, if none is specified */ if (pjsua_var.calls[0].med_tp == NULL) { pjsua_transport_config transport_cfg; /* Create default transport config */ pjsua_transport_config_default(&transport_cfg); transport_cfg.port = DEFAULT_RTP_PORT; status = pjsua_media_transports_create(&transport_cfg); if (status != PJ_SUCCESS) return status; } pj_timer_entry_init(&pjsua_var.snd_idle_timer, PJ_FALSE, NULL, &close_snd_timer_cb); return PJ_SUCCESS; } /* * Destroy pjsua media subsystem. */ pj_status_t pjsua_media_subsys_destroy(void) { unsigned i; close_snd_dev(); if (pjsua_var.mconf) { pjmedia_conf_destroy(pjsua_var.mconf); pjsua_var.mconf = NULL; } if (pjsua_var.null_port) { pjmedia_port_destroy(pjsua_var.null_port); pjsua_var.null_port = NULL; } /* Destroy file players */ for (i=0; iptr = host_port->ptr; host->slen = (str_port.ptr - host->ptr); str_port.ptr++; str_port.slen = host_port->slen - host->slen - 1; iport = (int)pj_strtoul(&str_port); if (iport < 1 || iport > 65535) return PJ_EINVAL; *port = (pj_uint16_t)iport; } else { *host = *host_port; *port = 0; } return PJ_SUCCESS; } /* Create ICE media transports (when ice is enabled) */ static pj_status_t create_ice_media_transports(void) { char stunip[PJ_INET6_ADDRSTRLEN]; pj_ice_strans_cfg ice_cfg; unsigned i; pj_status_t status; /* Make sure STUN server resolution has completed */ status = pjsua_resolve_stun_server(PJ_TRUE); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error resolving STUN server", status); return status; } /* Create ICE stream transport configuration */ pj_ice_strans_cfg_default(&ice_cfg); pj_stun_config_init(&ice_cfg.stun_cfg, &pjsua_var.cp.factory, 0, pjsip_endpt_get_ioqueue(pjsua_var.endpt), pjsip_endpt_get_timer_heap(pjsua_var.endpt)); ice_cfg.af = pj_AF_INET(); ice_cfg.resolver = pjsua_var.resolver; /* Configure STUN settings */ if (pj_sockaddr_has_addr(&pjsua_var.stun_srv)) { pj_sockaddr_print(&pjsua_var.stun_srv, stunip, sizeof(stunip), 0); ice_cfg.stun.server = pj_str(stunip); ice_cfg.stun.port = pj_sockaddr_get_port(&pjsua_var.stun_srv); } ice_cfg.stun.no_host_cands = pjsua_var.media_cfg.ice_no_host_cands; /* Configure TURN settings */ if (pjsua_var.media_cfg.enable_turn) { status = parse_host_port(&pjsua_var.media_cfg.turn_server, &ice_cfg.turn.server, &ice_cfg.turn.port); if (status != PJ_SUCCESS || ice_cfg.turn.server.slen == 0) { PJ_LOG(1,(THIS_FILE, "Invalid TURN server setting")); return PJ_EINVAL; } if (ice_cfg.turn.port == 0) ice_cfg.turn.port = 3479; ice_cfg.turn.conn_type = pjsua_var.media_cfg.turn_conn_type; pj_memcpy(&ice_cfg.turn.auth_cred, &pjsua_var.media_cfg.turn_auth_cred, sizeof(ice_cfg.turn.auth_cred)); } /* Create each media transport */ for (i=0; i0, PJ_EINVALIDOP); PJSUA_LOCK(); /* Delete existing media transports */ for (i=0; iacc_id]; pjmedia_srtp_setting srtp_opt; pjmedia_transport *srtp = NULL; #endif PJ_UNUSED_ARG(role); /* Return error if media transport has not been created yet * (e.g. application is starting) */ if (call->med_tp == NULL) { if (sip_err_code) *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR; return PJ_EBUSY; } #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) /* This function may be called when SRTP transport already exists * (e.g: in re-invite, update), don't need to destroy/re-create. */ if (!call->med_orig || call->med_tp == call->med_orig) { /* Check if SRTP requires secure signaling */ if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) { if (security_level < acc->cfg.srtp_secure_signaling) { if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE; return PJSIP_ESESSIONINSECURE; } } /* Always create SRTP adapter */ pjmedia_srtp_setting_default(&srtp_opt); srtp_opt.close_member_tp = PJ_FALSE; /* If media session has been ever established, let's use remote's * preference in SRTP usage policy, especially when it is stricter. */ if (call->rem_srtp_use > acc->cfg.use_srtp) srtp_opt.use = call->rem_srtp_use; else srtp_opt.use = acc->cfg.use_srtp; status = pjmedia_transport_srtp_create(pjsua_var.med_endpt, call->med_tp, &srtp_opt, &srtp); if (status != PJ_SUCCESS) { if (sip_err_code) *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR; return status; } /* Set SRTP as current media transport */ call->med_orig = call->med_tp; call->med_tp = srtp; } #else call->med_orig = call->med_tp; PJ_UNUSED_ARG(security_level); #endif /* Find out which media line in SDP that we support. If we are offerer, * audio will be at index 0 in SDP. */ if (rem_sdp == 0) { call->audio_idx = 0; } /* Otherwise find out the candidate audio media line in SDP */ else { unsigned i; pj_bool_t srtp_active; #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) srtp_active = acc->cfg.use_srtp && srtp != NULL; #else srtp_active = PJ_FALSE; #endif /* Media count must have been checked */ pj_assert(rem_sdp->media_count != 0); for (i=0; imedia_count; ++i) { const pjmedia_sdp_media *m = rem_sdp->media[i]; /* Skip if media is not audio */ if (pj_stricmp2(&m->desc.media, "audio") != 0) continue; /* Skip if media is disabled */ if (m->desc.port == 0) continue; /* Skip if transport is not supported */ if (pj_stricmp2(&m->desc.transport, "RTP/AVP") != 0 && pj_stricmp2(&m->desc.transport, "RTP/SAVP") != 0) { continue; } if (call->audio_idx == -1) { call->audio_idx = i; } else { /* We've found multiple candidates. This could happen * e.g. when remote is offering both RTP/AVP and RTP/AVP, * or when remote for some reason offers two audio. */ if (srtp_active && pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0) { /* Prefer RTP/SAVP when our media transport is SRTP */ call->audio_idx = i; } else if (!srtp_active && pj_stricmp2(&m->desc.transport, "RTP/AVP")==0) { /* Prefer RTP/AVP when our media transport is NOT SRTP */ call->audio_idx = i; } } } } /* Reject offer if we couldn't find a good m=audio line in offer */ if (call->audio_idx < 0) { if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE; pjsua_media_channel_deinit(call_id); return PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE_HERE); } PJ_LOG(4,(THIS_FILE, "Media index %d selected for call %d", call->audio_idx, call->index)); /* Create the media transport */ status = pjmedia_transport_media_create(call->med_tp, tmp_pool, 0, rem_sdp, call->audio_idx); if (status != PJ_SUCCESS) { if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE; pjsua_media_channel_deinit(call_id); return status; } call->med_tp_st = PJSUA_MED_TP_INIT; return PJ_SUCCESS; } pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id, pj_pool_t *pool, const pjmedia_sdp_session *rem_sdp, pjmedia_sdp_session **p_sdp, int *sip_status_code) { enum { MAX_MEDIA = 1 }; pjmedia_sdp_session *sdp; pjmedia_transport_info tpinfo; pjsua_call *call = &pjsua_var.calls[call_id]; pj_status_t status; /* Return error if media transport has not been created yet * (e.g. application is starting) */ if (call->med_tp == NULL) { return PJ_EBUSY; } /* Media index must have been determined before */ pj_assert(call->audio_idx != -1); /* Create media if it's not created. This could happen when call is * currently on-hold */ if (call->med_tp_st == PJSUA_MED_TP_IDLE) { pjsip_role_e role; role = (rem_sdp ? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC); status = pjsua_media_channel_init(call_id, role, call->secure_level, pool, rem_sdp, sip_status_code); if (status != PJ_SUCCESS) return status; } /* Get media socket info */ pjmedia_transport_info_init(&tpinfo); pjmedia_transport_get_info(call->med_tp, &tpinfo); /* Create SDP */ status = pjmedia_endpt_create_sdp(pjsua_var.med_endpt, pool, MAX_MEDIA, &tpinfo.sock_info, &sdp); if (status != PJ_SUCCESS) { if (sip_status_code) *sip_status_code = 500; return status; } /* If we're answering and the selected media is not the first media * in SDP, then fill in the unselected media with with zero port. * Otherwise we'll crash in transport_encode_sdp() because the media * lines are not aligned between offer and answer. */ if (rem_sdp && call->audio_idx != 0) { unsigned i; for (i=0; imedia_count; ++i) { const pjmedia_sdp_media *rem_m = rem_sdp->media[i]; pjmedia_sdp_media *m; const pjmedia_sdp_attr *a; if ((int)i == call->audio_idx) continue; m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media); pj_strdup(pool, &m->desc.media, &rem_m->desc.media); pj_strdup(pool, &m->desc.transport, &rem_m->desc.transport); m->desc.port = 0; /* Add one format, copy from the offer. And copy the corresponding * rtpmap and fmtp attributes too. */ m->desc.fmt_count = 1; pj_strdup(pool, &m->desc.fmt[0], &rem_m->desc.fmt[0]); if ((a=pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr, "rtpmap", &m->desc.fmt[0])) != NULL) { m->attr[m->attr_count++] = pjmedia_sdp_attr_clone(pool, a); } if ((a=pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr, "fmtp", &m->desc.fmt[0])) != NULL) { m->attr[m->attr_count++] = pjmedia_sdp_attr_clone(pool, a); } if (i==sdp->media_count) sdp->media[sdp->media_count++] = m; else { pj_array_insert(sdp->media, sizeof(sdp->media[0]), sdp->media_count, i, &m); ++sdp->media_count; } } } /* Add NAT info in the SDP */ if (pjsua_var.ua_cfg.nat_type_in_sdp) { pjmedia_sdp_attr *a; pj_str_t value; char nat_info[80]; value.ptr = nat_info; if (pjsua_var.ua_cfg.nat_type_in_sdp == 1) { value.slen = pj_ansi_snprintf(nat_info, sizeof(nat_info), "%d", pjsua_var.nat_type); } else { const char *type_name = pj_stun_get_nat_name(pjsua_var.nat_type); value.slen = pj_ansi_snprintf(nat_info, sizeof(nat_info), "%d %s", pjsua_var.nat_type, type_name); } a = pjmedia_sdp_attr_create(pool, "X-nat", &value); pjmedia_sdp_attr_add(&sdp->attr_count, sdp->attr, a); } /* Give the SDP to media transport */ status = pjmedia_transport_encode_sdp(call->med_tp, pool, sdp, rem_sdp, call->audio_idx); if (status != PJ_SUCCESS) { if (sip_status_code) *sip_status_code = PJSIP_SC_NOT_ACCEPTABLE; return status; } *p_sdp = sdp; return PJ_SUCCESS; } static void stop_media_session(pjsua_call_id call_id) { pjsua_call *call = &pjsua_var.calls[call_id]; if (call->conf_slot != PJSUA_INVALID_ID) { if (pjsua_var.mconf) { pjsua_conf_remove_port(call->conf_slot); } call->conf_slot = PJSUA_INVALID_ID; } if (call->session) { pjmedia_rtcp_stat stat; if (pjmedia_session_get_stream_stat(call->session, 0, &stat) == PJ_SUCCESS) { /* Save RTP timestamp & sequence, so when media session is * restarted, those values will be restored as the initial * RTP timestamp & sequence of the new media session. So in * the same call session, RTP timestamp and sequence are * guaranteed to be contigue. */ call->rtp_tx_seq_ts_set = 1 | (1 << 1); call->rtp_tx_seq = stat.rtp_tx_last_seq; call->rtp_tx_ts = stat.rtp_tx_last_ts; } if (pjsua_var.ua_cfg.cb.on_stream_destroyed) { pjsua_var.ua_cfg.cb.on_stream_destroyed(call_id, call->session, 0); } pjmedia_session_destroy(call->session); call->session = NULL; PJ_LOG(4,(THIS_FILE, "Media session for call %d is destroyed", call_id)); } call->media_st = PJSUA_CALL_MEDIA_NONE; } pj_status_t pjsua_media_channel_deinit(pjsua_call_id call_id) { pjsua_call *call = &pjsua_var.calls[call_id]; stop_media_session(call_id); if (call->med_tp_st != PJSUA_MED_TP_IDLE) { pjmedia_transport_media_stop(call->med_tp); call->med_tp_st = PJSUA_MED_TP_IDLE; } if (call->med_orig && call->med_tp && call->med_tp != call->med_orig) { pjmedia_transport_close(call->med_tp); call->med_tp = call->med_orig; } check_snd_dev_idle(); return PJ_SUCCESS; } /* * DTMF callback from the stream. */ static void dtmf_callback(pjmedia_stream *strm, void *user_data, int digit) { PJ_UNUSED_ARG(strm); /* For discussions about call mutex protection related to this * callback, please see ticket #460: * http://trac.pjsip.org/repos/ticket/460#comment:4 */ if (pjsua_var.ua_cfg.cb.on_dtmf_digit) { pjsua_call_id call_id; call_id = (pjsua_call_id)(long)user_data; pjsua_var.ua_cfg.cb.on_dtmf_digit(call_id, digit); } } pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, const pjmedia_sdp_session *local_sdp, const pjmedia_sdp_session *remote_sdp) { int prev_media_st = 0; pjsua_call *call = &pjsua_var.calls[call_id]; pjmedia_session_info sess_info; pjmedia_stream_info *si = NULL; pjmedia_port *media_port; pj_status_t status; /* Destroy existing media session, if any. */ prev_media_st = call->media_st; stop_media_session(call->index); /* Create media session info based on SDP parameters. */ status = pjmedia_session_info_from_sdp( call->inv->dlg->pool, pjsua_var.med_endpt, PJMEDIA_MAX_SDP_MEDIA, &sess_info, local_sdp, remote_sdp); if (status != PJ_SUCCESS) return status; /* Find which session is audio */ PJ_ASSERT_RETURN(call->audio_idx != -1, PJ_EBUG); PJ_ASSERT_RETURN(call->audio_idx < (int)sess_info.stream_cnt, PJ_EBUG); si = &sess_info.stream_info[call->audio_idx]; /* Reset session info with only one media stream */ sess_info.stream_cnt = 1; if (si != &sess_info.stream_info[0]) pj_memcpy(&sess_info.stream_info[0], si, sizeof(pjmedia_stream_info)); /* Check if no media is active */ if (sess_info.stream_cnt == 0 || si->dir == PJMEDIA_DIR_NONE) { /* Call media state */ call->media_st = PJSUA_CALL_MEDIA_NONE; /* Call media direction */ call->media_dir = PJMEDIA_DIR_NONE; /* Shutdown transport's session */ pjmedia_transport_media_stop(call->med_tp); call->med_tp_st = PJSUA_MED_TP_IDLE; /* No need because we need keepalive? */ /* Close upper entry of transport stack */ if (call->med_orig && (call->med_tp != call->med_orig)) { pjmedia_transport_close(call->med_tp); call->med_tp = call->med_orig; } } else { pjmedia_transport_info tp_info; /* Start/restart media transport */ status = pjmedia_transport_media_start(call->med_tp, call->inv->pool, local_sdp, remote_sdp, 0); if (status != PJ_SUCCESS) return status; call->med_tp_st = PJSUA_MED_TP_RUNNING; /* Get remote SRTP usage policy */ pjmedia_transport_info_init(&tp_info); pjmedia_transport_get_info(call->med_tp, &tp_info); if (tp_info.specific_info_cnt > 0) { int i; for (i = 0; i < tp_info.specific_info_cnt; ++i) { if (tp_info.spc_info[i].type == PJMEDIA_TRANSPORT_TYPE_SRTP) { pjmedia_srtp_info *srtp_info = (pjmedia_srtp_info*) tp_info.spc_info[i].buffer; call->rem_srtp_use = srtp_info->peer_use; break; } } } /* Override ptime, if this option is specified. */ if (pjsua_var.media_cfg.ptime != 0) { si->param->setting.frm_per_pkt = (pj_uint8_t) (pjsua_var.media_cfg.ptime / si->param->info.frm_ptime); if (si->param->setting.frm_per_pkt == 0) si->param->setting.frm_per_pkt = 1; } /* Disable VAD, if this option is specified. */ if (pjsua_var.media_cfg.no_vad) { si->param->setting.vad = 0; } /* Optionally, application may modify other stream settings here * (such as jitter buffer parameters, codec ptime, etc.) */ si->jb_init = pjsua_var.media_cfg.jb_init; si->jb_min_pre = pjsua_var.media_cfg.jb_min_pre; si->jb_max_pre = pjsua_var.media_cfg.jb_max_pre; si->jb_max = pjsua_var.media_cfg.jb_max; /* Set SSRC */ si->ssrc = call->ssrc; /* Set RTP timestamp & sequence, normally these value are intialized * automatically when stream session created, but for some cases (e.g: * call reinvite, call update) timestamp and sequence need to be kept * contigue. */ si->rtp_ts = call->rtp_tx_ts; si->rtp_seq = call->rtp_tx_seq; si->rtp_seq_ts_set = call->rtp_tx_seq_ts_set; /* Create session based on session info. */ status = pjmedia_session_create( pjsua_var.med_endpt, &sess_info, &call->med_tp, call, &call->session ); if (status != PJ_SUCCESS) { return status; } /* If DTMF callback is installed by application, install our * callback to the session. */ if (pjsua_var.ua_cfg.cb.on_dtmf_digit) { pjmedia_session_set_dtmf_callback(call->session, 0, &dtmf_callback, (void*)(long)(call->index)); } /* Get the port interface of the first stream in the session. * We need the port interface to add to the conference bridge. */ pjmedia_session_get_port(call->session, 0, &media_port); /* Notify application about stream creation. * Note: application may modify media_port to point to different * media port */ if (pjsua_var.ua_cfg.cb.on_stream_created) { pjsua_var.ua_cfg.cb.on_stream_created(call_id, call->session, 0, &media_port); } /* * Add the call to conference bridge. */ { char tmp[PJSIP_MAX_URL_SIZE]; pj_str_t port_name; port_name.ptr = tmp; port_name.slen = pjsip_uri_print(PJSIP_URI_IN_REQ_URI, call->inv->dlg->remote.info->uri, tmp, sizeof(tmp)); if (port_name.slen < 1) { port_name = pj_str("call"); } status = pjmedia_conf_add_port( pjsua_var.mconf, call->inv->pool, media_port, &port_name, (unsigned*)&call->conf_slot); if (status != PJ_SUCCESS) { return status; } } /* Call media direction */ call->media_dir = si->dir; /* Call media state */ if (call->local_hold) call->media_st = PJSUA_CALL_MEDIA_LOCAL_HOLD; else if (call->media_dir == PJMEDIA_DIR_DECODING) call->media_st = PJSUA_CALL_MEDIA_REMOTE_HOLD; else call->media_st = PJSUA_CALL_MEDIA_ACTIVE; } /* Print info. */ { char info[80]; int info_len = 0; unsigned i; for (i=0; idir) { case PJMEDIA_DIR_NONE: dir = "inactive"; break; case PJMEDIA_DIR_ENCODING: dir = "sendonly"; break; case PJMEDIA_DIR_DECODING: dir = "recvonly"; break; case PJMEDIA_DIR_ENCODING_DECODING: dir = "sendrecv"; break; default: dir = "unknown"; break; } len = pj_ansi_sprintf( info+info_len, ", stream #%d: %.*s (%s)", i, (int)strm_info->fmt.encoding_name.slen, strm_info->fmt.encoding_name.ptr, dir); if (len > 0) info_len += len; } PJ_LOG(4,(THIS_FILE,"Media updates%s", info)); } return PJ_SUCCESS; } /* * Get maxinum number of conference ports. */ PJ_DEF(unsigned) pjsua_conf_get_max_ports(void) { return pjsua_var.media_cfg.max_media_ports; } /* * Get current number of active ports in the bridge. */ PJ_DEF(unsigned) pjsua_conf_get_active_ports(void) { unsigned ports[PJSUA_MAX_CONF_PORTS]; unsigned count = PJ_ARRAY_SIZE(ports); pj_status_t status; status = pjmedia_conf_enum_ports(pjsua_var.mconf, ports, &count); if (status != PJ_SUCCESS) count = 0; return count; } /* * Enumerate all conference ports. */ PJ_DEF(pj_status_t) pjsua_enum_conf_ports(pjsua_conf_port_id id[], unsigned *count) { return pjmedia_conf_enum_ports(pjsua_var.mconf, (unsigned*)id, count); } /* * Get information about the specified conference port */ PJ_DEF(pj_status_t) pjsua_conf_get_port_info( pjsua_conf_port_id id, pjsua_conf_port_info *info) { pjmedia_conf_port_info cinfo; unsigned i; pj_status_t status; status = pjmedia_conf_get_port_info( pjsua_var.mconf, id, &cinfo); if (status != PJ_SUCCESS) return status; pj_bzero(info, sizeof(*info)); info->slot_id = id; info->name = cinfo.name; info->clock_rate = cinfo.clock_rate; info->channel_count = cinfo.channel_count; info->samples_per_frame = cinfo.samples_per_frame; info->bits_per_sample = cinfo.bits_per_sample; /* Build array of listeners */ info->listener_cnt = cinfo.listener_cnt; for (i=0; ilisteners[i] = cinfo.listener_slots[i]; } return PJ_SUCCESS; } /* * Add arbitrary media port to PJSUA's conference bridge. */ PJ_DEF(pj_status_t) pjsua_conf_add_port( pj_pool_t *pool, pjmedia_port *port, pjsua_conf_port_id *p_id) { pj_status_t status; status = pjmedia_conf_add_port(pjsua_var.mconf, pool, port, NULL, (unsigned*)p_id); if (status != PJ_SUCCESS) { if (p_id) *p_id = PJSUA_INVALID_ID; } return status; } /* * Remove arbitrary slot from the conference bridge. */ PJ_DEF(pj_status_t) pjsua_conf_remove_port(pjsua_conf_port_id id) { pj_status_t status; status = pjmedia_conf_remove_port(pjsua_var.mconf, (unsigned)id); check_snd_dev_idle(); return status; } /* * Establish unidirectional media flow from souce to sink. */ PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, pjsua_conf_port_id sink) { /* If sound device idle timer is active, cancel it first. */ PJSUA_LOCK(); if (pjsua_var.snd_idle_timer.id) { pjsip_endpt_cancel_timer(pjsua_var.endpt, &pjsua_var.snd_idle_timer); pjsua_var.snd_idle_timer.id = PJ_FALSE; } PJSUA_UNLOCK(); /* Create sound port if none is instantiated */ if (pjsua_var.snd_port==NULL && pjsua_var.null_snd==NULL && !pjsua_var.no_snd) { pj_status_t status; status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error opening sound device", status); return status; } } return pjmedia_conf_connect_port(pjsua_var.mconf, source, sink, 0); } /* * Disconnect media flow from the source to destination port. */ PJ_DEF(pj_status_t) pjsua_conf_disconnect( pjsua_conf_port_id source, pjsua_conf_port_id sink) { pj_status_t status; status = pjmedia_conf_disconnect_port(pjsua_var.mconf, source, sink); check_snd_dev_idle(); return status; } /* * Adjust the signal level to be transmitted from the bridge to the * specified port by making it louder or quieter. */ PJ_DEF(pj_status_t) pjsua_conf_adjust_tx_level(pjsua_conf_port_id slot, float level) { return pjmedia_conf_adjust_tx_level(pjsua_var.mconf, slot, (int)((level-1) * 128)); } /* * Adjust the signal level to be received from the specified port (to * the bridge) by making it louder or quieter. */ PJ_DEF(pj_status_t) pjsua_conf_adjust_rx_level(pjsua_conf_port_id slot, float level) { return pjmedia_conf_adjust_rx_level(pjsua_var.mconf, slot, (int)((level-1) * 128)); } /* * Get last signal level transmitted to or received from the specified port. */ PJ_DEF(pj_status_t) pjsua_conf_get_signal_level(pjsua_conf_port_id slot, unsigned *tx_level, unsigned *rx_level) { return pjmedia_conf_get_signal_level(pjsua_var.mconf, slot, tx_level, rx_level); } /***************************************************************************** * File player. */ static char* get_basename(const char *path, unsigned len) { char *p = ((char*)path) + len; if (len==0) return p; for (--p; p!=path && *p!='/' && *p!='\\'; ) --p; return (p==path) ? p : p+1; } /* * Create a file player, and automatically connect this player to * the conference bridge. */ PJ_DEF(pj_status_t) pjsua_player_create( const pj_str_t *filename, unsigned options, pjsua_player_id *p_id) { unsigned slot, file_id; char path[PJ_MAXPATH]; pj_pool_t *pool; pjmedia_port *port; pj_status_t status; if (pjsua_var.player_cnt >= PJ_ARRAY_SIZE(pjsua_var.player)) return PJ_ETOOMANY; PJSUA_LOCK(); for (file_id=0; file_idptr, filename->slen); path[filename->slen] = '\0'; pool = pjsua_pool_create(get_basename(path, filename->slen), 1000, 1000); if (!pool) { PJSUA_UNLOCK(); return PJ_ENOMEM; } status = pjmedia_wav_player_port_create( pool, path, pjsua_var.mconf_cfg.samples_per_frame * 1000 / pjsua_var.media_cfg.channel_count / pjsua_var.media_cfg.clock_rate, options, 0, &port); if (status != PJ_SUCCESS) { PJSUA_UNLOCK(); pjsua_perror(THIS_FILE, "Unable to open file for playback", status); pj_pool_release(pool); return status; } status = pjmedia_conf_add_port(pjsua_var.mconf, pool, port, filename, &slot); if (status != PJ_SUCCESS) { pjmedia_port_destroy(port); PJSUA_UNLOCK(); pjsua_perror(THIS_FILE, "Unable to add file to conference bridge", status); pj_pool_release(pool); return status; } pjsua_var.player[file_id].type = 0; pjsua_var.player[file_id].pool = pool; pjsua_var.player[file_id].port = port; pjsua_var.player[file_id].slot = slot; if (p_id) *p_id = file_id; ++pjsua_var.player_cnt; PJSUA_UNLOCK(); return PJ_SUCCESS; } /* * Create a file playlist media port, and automatically add the port * to the conference bridge. */ PJ_DEF(pj_status_t) pjsua_playlist_create( const pj_str_t file_names[], unsigned file_count, const pj_str_t *label, unsigned options, pjsua_player_id *p_id) { unsigned slot, file_id, ptime; pj_pool_t *pool; pjmedia_port *port; pj_status_t status; if (pjsua_var.player_cnt >= PJ_ARRAY_SIZE(pjsua_var.player)) return PJ_ETOOMANY; PJSUA_LOCK(); for (file_id=0; file_idinfo.name, &slot); if (status != PJ_SUCCESS) { pjmedia_port_destroy(port); PJSUA_UNLOCK(); pjsua_perror(THIS_FILE, "Unable to add port", status); pj_pool_release(pool); return status; } pjsua_var.player[file_id].type = 1; pjsua_var.player[file_id].pool = pool; pjsua_var.player[file_id].port = port; pjsua_var.player[file_id].slot = slot; if (p_id) *p_id = file_id; ++pjsua_var.player_cnt; PJSUA_UNLOCK(); return PJ_SUCCESS; } /* * Get conference port ID associated with player. */ PJ_DEF(pjsua_conf_port_id) pjsua_player_get_conf_port(pjsua_player_id id) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); return pjsua_var.player[id].slot; } /* * Get the media port for the player. */ PJ_DEF(pj_status_t) pjsua_player_get_port( pjsua_player_id id, pjmedia_port **p_port) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(p_port != NULL, PJ_EINVAL); *p_port = pjsua_var.player[id].port; return PJ_SUCCESS; } /* * Set playback position. */ PJ_DEF(pj_status_t) pjsua_player_set_pos( pjsua_player_id id, pj_uint32_t samples) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].type == 0, PJ_EINVAL); return pjmedia_wav_player_port_set_pos(pjsua_var.player[id].port, samples); } /* * Close the file, remove the player from the bridge, and free * resources associated with the file player. */ PJ_DEF(pj_status_t) pjsua_player_destroy(pjsua_player_id id) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJSUA_LOCK(); if (pjsua_var.player[id].port) { pjsua_conf_remove_port(pjsua_var.player[id].slot); pjmedia_port_destroy(pjsua_var.player[id].port); pjsua_var.player[id].port = NULL; pjsua_var.player[id].slot = 0xFFFF; pj_pool_release(pjsua_var.player[id].pool); pjsua_var.player[id].pool = NULL; pjsua_var.player_cnt--; } PJSUA_UNLOCK(); return PJ_SUCCESS; } /***************************************************************************** * File recorder. */ /* * Create a file recorder, and automatically connect this recorder to * the conference bridge. */ PJ_DEF(pj_status_t) pjsua_recorder_create( const pj_str_t *filename, unsigned enc_type, void *enc_param, pj_ssize_t max_size, unsigned options, pjsua_recorder_id *p_id) { enum Format { FMT_UNKNOWN, FMT_WAV, FMT_MP3, }; unsigned slot, file_id; char path[PJ_MAXPATH]; pj_str_t ext; int file_format; pj_pool_t *pool; pjmedia_port *port; pj_status_t status; /* Filename must present */ PJ_ASSERT_RETURN(filename != NULL, PJ_EINVAL); /* Don't support max_size at present */ PJ_ASSERT_RETURN(max_size == 0 || max_size == -1, PJ_EINVAL); /* Don't support encoding type at present */ PJ_ASSERT_RETURN(enc_type == 0, PJ_EINVAL); if (pjsua_var.rec_cnt >= PJ_ARRAY_SIZE(pjsua_var.recorder)) return PJ_ETOOMANY; /* Determine the file format */ ext.ptr = filename->ptr + filename->slen - 4; ext.slen = 4; if (pj_stricmp2(&ext, ".wav") == 0) file_format = FMT_WAV; else if (pj_stricmp2(&ext, ".mp3") == 0) file_format = FMT_MP3; else { PJ_LOG(1,(THIS_FILE, "pjsua_recorder_create() error: unable to " "determine file format for %.*s", (int)filename->slen, filename->ptr)); return PJ_ENOTSUP; } PJSUA_LOCK(); for (file_id=0; file_idptr, filename->slen); path[filename->slen] = '\0'; pool = pjsua_pool_create(get_basename(path, filename->slen), 1000, 1000); if (!pool) { PJSUA_UNLOCK(); return PJ_ENOMEM; } if (file_format == FMT_WAV) { status = pjmedia_wav_writer_port_create(pool, path, pjsua_var.media_cfg.clock_rate, pjsua_var.mconf_cfg.channel_count, pjsua_var.mconf_cfg.samples_per_frame, pjsua_var.mconf_cfg.bits_per_sample, options, 0, &port); } else { PJ_UNUSED_ARG(enc_param); port = NULL; status = PJ_ENOTSUP; } if (status != PJ_SUCCESS) { PJSUA_UNLOCK(); pjsua_perror(THIS_FILE, "Unable to open file for recording", status); pj_pool_release(pool); return status; } status = pjmedia_conf_add_port(pjsua_var.mconf, pool, port, filename, &slot); if (status != PJ_SUCCESS) { pjmedia_port_destroy(port); PJSUA_UNLOCK(); pj_pool_release(pool); return status; } pjsua_var.recorder[file_id].port = port; pjsua_var.recorder[file_id].slot = slot; pjsua_var.recorder[file_id].pool = pool; if (p_id) *p_id = file_id; ++pjsua_var.rec_cnt; PJSUA_UNLOCK(); return PJ_SUCCESS; } /* * Get conference port associated with recorder. */ PJ_DEF(pjsua_conf_port_id) pjsua_recorder_get_conf_port(pjsua_recorder_id id) { PJ_ASSERT_RETURN(id>=0 && id<(int)PJ_ARRAY_SIZE(pjsua_var.recorder), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.recorder[id].port != NULL, PJ_EINVAL); return pjsua_var.recorder[id].slot; } /* * Get the media port for the recorder. */ PJ_DEF(pj_status_t) pjsua_recorder_get_port( pjsua_recorder_id id, pjmedia_port **p_port) { PJ_ASSERT_RETURN(id>=0 && id<(int)PJ_ARRAY_SIZE(pjsua_var.recorder), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.recorder[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(p_port != NULL, PJ_EINVAL); *p_port = pjsua_var.recorder[id].port; return PJ_SUCCESS; } /* * Destroy recorder (this will complete recording). */ PJ_DEF(pj_status_t) pjsua_recorder_destroy(pjsua_recorder_id id) { PJ_ASSERT_RETURN(id>=0 && id<(int)PJ_ARRAY_SIZE(pjsua_var.recorder), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.recorder[id].port != NULL, PJ_EINVAL); PJSUA_LOCK(); if (pjsua_var.recorder[id].port) { pjsua_conf_remove_port(pjsua_var.recorder[id].slot); pjmedia_port_destroy(pjsua_var.recorder[id].port); pjsua_var.recorder[id].port = NULL; pjsua_var.recorder[id].slot = 0xFFFF; pj_pool_release(pjsua_var.recorder[id].pool); pjsua_var.recorder[id].pool = NULL; pjsua_var.rec_cnt--; } PJSUA_UNLOCK(); return PJ_SUCCESS; } /***************************************************************************** * Sound devices. */ /* * Enum sound devices. */ PJ_DEF(pj_status_t) pjsua_enum_snd_devs( pjmedia_snd_dev_info info[], unsigned *count) { unsigned i, dev_count; dev_count = pjmedia_snd_get_dev_count(); if (dev_count > *count) dev_count = *count; for (i=0; iname, cap_info->name)); pjmedia_snd_port_disconnect(pjsua_var.snd_port); pjmedia_snd_port_destroy(pjsua_var.snd_port); pjsua_var.snd_port = NULL; } /* Close null sound device */ if (pjsua_var.null_snd) { PJ_LOG(4,(THIS_FILE, "Closing null sound device..")); pjmedia_master_port_destroy(pjsua_var.null_snd, PJ_FALSE); pjsua_var.null_snd = NULL; } if (pjsua_var.snd_pool) pj_pool_release(pjsua_var.snd_pool); pjsua_var.snd_pool = NULL; } /* * Select or change sound device. Application may call this function at * any time to replace current sound device. */ PJ_DEF(pj_status_t) pjsua_set_snd_dev( int capture_dev, int playback_dev) { pjmedia_port *conf_port; const pjmedia_snd_dev_info *play_info; unsigned clock_rates[] = {0, 44100, 48000, 32000, 16000, 8000}; unsigned selected_clock_rate = 0; unsigned i; pjmedia_snd_stream *strm; pjmedia_snd_stream_info si; pj_str_t tmp; pj_status_t status = -1; /* Check if NULL sound device is used */ if (NULL_SND_DEV_ID == capture_dev || NULL_SND_DEV_ID == playback_dev) { return pjsua_set_null_snd_dev(); } /* Close existing sound port */ close_snd_dev(); /* Create memory pool for sound device. */ pjsua_var.snd_pool = pjsua_pool_create("pjsua_snd", 4000, 4000); PJ_ASSERT_RETURN(pjsua_var.snd_pool, PJ_ENOMEM); /* Set default clock rate */ clock_rates[0] = pjsua_var.media_cfg.snd_clock_rate; if (clock_rates[0] == 0) clock_rates[0] = pjsua_var.media_cfg.clock_rate; /* Get the port0 of the conference bridge. */ conf_port = pjmedia_conf_get_master_port(pjsua_var.mconf); pj_assert(conf_port != NULL); /* Attempts to open the sound device with different clock rates */ for (i=0; i= 3 && pjsua_var.media_cfg.quality <= 4) { resample_opt |= PJMEDIA_RESAMPLE_USE_SMALL_FILTER; } else if (pjsua_var.media_cfg.quality < 3) { resample_opt |= PJMEDIA_RESAMPLE_USE_LINEAR; } status = pjmedia_resample_port_create(pjsua_var.snd_pool, conf_port, selected_clock_rate, resample_opt, &resample_port); if (status != PJ_SUCCESS) { pj_strerror(status, errmsg, sizeof(errmsg)); PJ_LOG(4, (THIS_FILE, "Error creating resample port, trying next " "clock rate", errmsg)); pjmedia_snd_port_destroy(pjsua_var.snd_port); pjsua_var.snd_port = NULL; continue; } else { conf_port = resample_port; break; } } else { break; } } pj_strerror(status, errmsg, sizeof(errmsg)); PJ_LOG(4, (THIS_FILE, "..failed: %s", errmsg)); } if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to open sound device", status); return status; } /* Set AEC */ pjmedia_snd_port_set_ec( pjsua_var.snd_port, pjsua_var.snd_pool, pjsua_var.media_cfg.ec_tail_len, pjsua_var.media_cfg.ec_options); /* Connect sound port to the bridge */ status = pjmedia_snd_port_connect(pjsua_var.snd_port, conf_port ); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to connect conference port to " "sound device", status); pjmedia_snd_port_destroy(pjsua_var.snd_port); pjsua_var.snd_port = NULL; return status; } /* Save the device IDs */ pjsua_var.cap_dev = capture_dev; pjsua_var.play_dev = playback_dev; /* Update sound device name. */ strm = pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port); pjmedia_snd_stream_get_info(strm, &si); play_info = pjmedia_snd_get_dev_info(si.rec_id); if (si.clock_rate != pjsua_var.media_cfg.clock_rate) { char tmp_buf[128]; int tmp_buf_len = sizeof(tmp_buf); tmp_buf_len = pj_ansi_snprintf(tmp_buf, sizeof(tmp_buf)-1, "%s (%dKHz)", play_info->name, si.clock_rate/1000); pj_strset(&tmp, tmp_buf, tmp_buf_len); pjmedia_conf_set_port0_name(pjsua_var.mconf, &tmp); } else { pjmedia_conf_set_port0_name(pjsua_var.mconf, pj_cstr(&tmp, play_info->name)); } return PJ_SUCCESS; } /* * Get currently active sound devices. If sound devices has not been created * (for example when pjsua_start() is not called), it is possible that * the function returns PJ_SUCCESS with -1 as device IDs. */ PJ_DEF(pj_status_t) pjsua_get_snd_dev(int *capture_dev, int *playback_dev) { if (capture_dev) { *capture_dev = pjsua_var.cap_dev; } if (playback_dev) { *playback_dev = pjsua_var.play_dev; } return PJ_SUCCESS; } /* * Use null sound device. */ PJ_DEF(pj_status_t) pjsua_set_null_snd_dev(void) { pjmedia_port *conf_port; pj_status_t status; /* Close existing sound device */ close_snd_dev(); /* Create memory pool for sound device. */ pjsua_var.snd_pool = pjsua_pool_create("pjsua_snd", 4000, 4000); PJ_ASSERT_RETURN(pjsua_var.snd_pool, PJ_ENOMEM); PJ_LOG(4,(THIS_FILE, "Opening null sound device..")); /* Get the port0 of the conference bridge. */ conf_port = pjmedia_conf_get_master_port(pjsua_var.mconf); pj_assert(conf_port != NULL); /* Create master port, connecting port0 of the conference bridge to * a null port. */ status = pjmedia_master_port_create(pjsua_var.snd_pool, pjsua_var.null_port, conf_port, 0, &pjsua_var.null_snd); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create null sound device", status); return status; } /* Start the master port */ status = pjmedia_master_port_start(pjsua_var.null_snd); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); pjsua_var.cap_dev = NULL_SND_DEV_ID; pjsua_var.play_dev = NULL_SND_DEV_ID; return PJ_SUCCESS; } /* * Use no device! */ PJ_DEF(pjmedia_port*) pjsua_set_no_snd_dev(void) { /* Close existing sound device */ close_snd_dev(); pjsua_var.no_snd = PJ_TRUE; return pjmedia_conf_get_master_port(pjsua_var.mconf); } /* * Configure the AEC settings of the sound port. */ PJ_DEF(pj_status_t) pjsua_set_ec(unsigned tail_ms, unsigned options) { pjsua_var.media_cfg.ec_tail_len = tail_ms; if (pjsua_var.snd_port) return pjmedia_snd_port_set_ec( pjsua_var.snd_port, pjsua_var.pool, tail_ms, options); return PJ_SUCCESS; } /* * Get current AEC tail length. */ PJ_DEF(pj_status_t) pjsua_get_ec_tail(unsigned *p_tail_ms) { *p_tail_ms = pjsua_var.media_cfg.ec_tail_len; return PJ_SUCCESS; } /***************************************************************************** * Codecs. */ /* * Enum all supported codecs in the system. */ PJ_DEF(pj_status_t) pjsua_enum_codecs( pjsua_codec_info id[], unsigned *p_count ) { pjmedia_codec_mgr *codec_mgr; pjmedia_codec_info info[32]; unsigned i, count, prio[32]; pj_status_t status; codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt); count = PJ_ARRAY_SIZE(info); status = pjmedia_codec_mgr_enum_codecs( codec_mgr, &count, info, prio); if (status != PJ_SUCCESS) { *p_count = 0; return status; } if (count > *p_count) count = *p_count; for (i=0; islen==1 && *codec_id->ptr=='*') codec_id = &all; return pjmedia_codec_mgr_set_codec_priority(codec_mgr, codec_id, priority); } /* * Get codec parameters. */ PJ_DEF(pj_status_t) pjsua_codec_get_param( const pj_str_t *codec_id, pjmedia_codec_param *param ) { const pj_str_t all = { NULL, 0 }; const pjmedia_codec_info *info; pjmedia_codec_mgr *codec_mgr; unsigned count = 1; pj_status_t status; codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt); if (codec_id->slen==1 && *codec_id->ptr=='*') codec_id = &all; status = pjmedia_codec_mgr_find_codecs_by_id(codec_mgr, codec_id, &count, &info, NULL); if (status != PJ_SUCCESS) return status; if (count != 1) return PJ_ENOTFOUND; status = pjmedia_codec_mgr_get_default_param( codec_mgr, info, param); return status; } /* * Set codec parameters. */ PJ_DEF(pj_status_t) pjsua_codec_set_param( const pj_str_t *id, const pjmedia_codec_param *param) { PJ_UNUSED_ARG(id); PJ_UNUSED_ARG(param); PJ_TODO(set_codec_param); return PJ_SUCCESS; }