diff options
Diffstat (limited to 'pjsip/src/pjsua-lib/pjsua_media.c')
| -rw-r--r-- | pjsip/src/pjsua-lib/pjsua_media.c | 1933 |
1 files changed, 1152 insertions, 781 deletions
diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c index 2eced3d9..94e81566 100644 --- a/pjsip/src/pjsua-lib/pjsua_media.c +++ b/pjsip/src/pjsua-lib/pjsua_media.c @@ -58,6 +58,7 @@ static void pjsua_media_config_dup(pj_pool_t *pool, pj_stun_auth_cred_dup(pool, &dst->turn_auth_cred, &src->turn_auth_cred); } + /** * Init media subsystems. */ @@ -65,6 +66,7 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) { pj_str_t codec_id = {NULL, 0}; unsigned opt; + pjmedia_audio_codec_config codec_cfg; pj_status_t status; /* To suppress warning about unused var when all codecs are disabled */ @@ -111,86 +113,13 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) 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 */ + /* + * Register all codecs + */ + pjmedia_audio_codec_config_default(&codec_cfg); + codec_cfg.speex.quality = pjsua_var.media_cfg.quality; + codec_cfg.speex.complexity = -1; + codec_cfg.ilbc.mode = pjsua_var.media_cfg.ilbc_mode; #if PJMEDIA_HAS_PASSTHROUGH_CODECS /* Register passthrough codecs */ @@ -198,7 +127,6 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) unsigned aud_idx; unsigned ext_fmt_cnt = 0; pjmedia_format ext_fmts[32]; - pjmedia_codec_passthrough_setting setting; /* List extended formats supported by audio devices */ for (aud_idx = 0; aud_idx < pjmedia_aud_dev_count(); ++aud_idx) { @@ -235,36 +163,31 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) } /* Init the passthrough codec with supported formats only */ - setting.fmt_cnt = ext_fmt_cnt; - setting.fmts = ext_fmts; - setting.ilbc_mode = cfg->ilbc_mode; - status = pjmedia_codec_passthrough_init2(pjsua_var.med_endpt, &setting); - if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Error initializing passthrough codecs", - status); - return status; - } + codec_cfg.passthrough.setting.fmt_cnt = ext_fmt_cnt; + codec_cfg.passthrough.setting.fmts = ext_fmts; + codec_cfg.passthrough.setting.ilbc_mode = cfg->ilbc_mode; } #endif /* PJMEDIA_HAS_PASSTHROUGH_CODECS */ -#if PJMEDIA_HAS_G7221_CODEC - /* Register G722.1 codecs */ - status = pjmedia_codec_g7221_init(pjsua_var.med_endpt); + /* Register all codecs */ + status = pjmedia_codec_register_audio_codecs(pjsua_var.med_endpt, + &codec_cfg); if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Error initializing G722.1 codec", - status); + PJ_PERROR(1,(THIS_FILE, status, "Error registering codecs")); return status; } -#endif /* PJMEDIA_HAS_G7221_CODEC */ -#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; - } + /* 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); /* Disable ALL L16 codecs */ codec_id = pj_str("L16"); @@ -272,8 +195,6 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) 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. @@ -296,7 +217,6 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) opt |= PJMEDIA_CONF_USE_LINEAR; } - /* Init conference bridge. */ status = pjmedia_conf_create(pjsua_var.pool, pjsua_var.media_cfg.max_media_ports, @@ -334,18 +254,214 @@ pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg) } #endif + /* Video */ +#if PJMEDIA_HAS_VIDEO + status = pjsua_vid_subsys_init(); + if (status != PJ_SUCCESS) + return status; +#endif + return PJ_SUCCESS; } -/* +/* Check if sound device is idle. */ +static void check_snd_dev_idle() +{ + unsigned call_cnt; + + /* Get the call count, we shouldn't close the sound device when there is + * any calls active. + */ + call_cnt = pjsua_call_get_count(); + + /* When this function is called from pjsua_media_channel_deinit() upon + * disconnecting call, actually the call count hasn't been updated/ + * decreased. So we put additional check here, if there is only one + * call and it's in DISCONNECTED state, there is actually no active + * call. + */ + if (call_cnt == 1) { + pjsua_call_id call_id; + pj_status_t status; + + status = pjsua_enum_calls(&call_id, &call_cnt); + if (status == PJ_SUCCESS && call_cnt > 0 && + !pjsua_call_is_active(call_id)) + { + call_cnt = 0; + } + } + + /* Activate sound device auto-close timer if sound device is idle. + * It is idle when there is no port connection in the bridge and + * there is no active call. + * + * Note: this block is now valid if no snd dev is used because of #1299 + */ + if ((pjsua_var.snd_port!=NULL || pjsua_var.null_snd!=NULL || + pjsua_var.no_snd) && + pjsua_var.snd_idle_timer.id == PJ_FALSE && + pjmedia_conf_get_connect_count(pjsua_var.mconf) == 0 && + call_cnt == 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; + +#if DISABLED_FOR_TICKET_1185 + /* Create media for calls, if none is specified */ + if (pjsua_var.calls[0].media[0].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; + } +#endif + + pj_timer_entry_init(&pjsua_var.snd_idle_timer, PJ_FALSE, NULL, + &close_snd_timer_cb); + + /* Video */ +#if PJMEDIA_HAS_VIDEO + status = pjsua_vid_subsys_start(); + if (status != PJ_SUCCESS) + return status; +#endif + + /* Perform NAT detection */ + status = pjsua_detect_nat_type(); + if (status != PJ_SUCCESS) { + PJ_PERROR(1,(THIS_FILE, status, "NAT type detection failed")); + } + + return PJ_SUCCESS; +} + + +/* + * Destroy pjsua media subsystem. + */ +pj_status_t pjsua_media_subsys_destroy(void) +{ + unsigned i; + + PJ_LOG(4,(THIS_FILE, "Shutting down media..")); + + 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; i<PJ_ARRAY_SIZE(pjsua_var.player); ++i) { + if (pjsua_var.player[i].port) { + pjmedia_port_destroy(pjsua_var.player[i].port); + pjsua_var.player[i].port = NULL; + } + } + + /* Destroy file recorders */ + for (i=0; i<PJ_ARRAY_SIZE(pjsua_var.recorder); ++i) { + if (pjsua_var.recorder[i].port) { + pjmedia_port_destroy(pjsua_var.recorder[i].port); + pjsua_var.recorder[i].port = NULL; + } + } + + /* Close media transports */ + for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { + unsigned strm_idx; + pjsua_call *call = &pjsua_var.calls[i]; + for (strm_idx=0; strm_idx<call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + if (call_med->tp_st != PJSUA_MED_TP_IDLE) { + pjsua_media_channel_deinit(i); + } + if (call_med->tp && call_med->tp_auto_del) { + pjmedia_transport_close(call_med->tp); + } + call_med->tp = NULL; + } + } + + /* Destroy media endpoint. */ + if (pjsua_var.med_endpt) { + +# if PJMEDIA_HAS_VIDEO + pjsua_vid_subsys_destroy(); +# endif + + pjmedia_endpt_destroy(pjsua_var.med_endpt); + pjsua_var.med_endpt = NULL; + + /* Deinitialize sound subsystem */ + // Not necessary, as pjmedia_snd_deinit() should have been called + // in pjmedia_endpt_destroy(). + //pjmedia_snd_deinit(); + } + + /* Reset RTP port */ + next_rtp_port = 0; + + 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 { + enum { RTP_RETRY = 100 }; int i; @@ -365,6 +481,9 @@ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, if (next_rtp_port == 0) next_rtp_port = (pj_uint16_t)cfg->port; + if (next_rtp_port == 0) + next_rtp_port = (pj_uint16_t)40000; + for (i=0; i<2; ++i) sock[i] = PJ_INVALID_SOCKET; @@ -389,15 +508,15 @@ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, } /* Apply QoS to RTP socket, if specified */ - status = pj_sock_apply_qos2(sock[0], cfg->qos_type, - &cfg->qos_params, + status = pj_sock_apply_qos2(sock[0], cfg->qos_type, + &cfg->qos_params, 2, THIS_FILE, "RTP socket"); /* Bind RTP socket */ - status=pj_sock_bind_in(sock[0], pj_ntohl(bound_addr.sin_addr.s_addr), + status=pj_sock_bind_in(sock[0], pj_ntohl(bound_addr.sin_addr.s_addr), next_rtp_port); if (status != PJ_SUCCESS) { - pj_sock_close(sock[0]); + pj_sock_close(sock[0]); sock[0] = PJ_INVALID_SOCKET; continue; } @@ -411,18 +530,18 @@ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, } /* Apply QoS to RTCP socket, if specified */ - status = pj_sock_apply_qos2(sock[1], cfg->qos_type, - &cfg->qos_params, + status = pj_sock_apply_qos2(sock[1], cfg->qos_type, + &cfg->qos_params, 2, THIS_FILE, "RTCP socket"); /* Bind RTCP socket */ - status=pj_sock_bind_in(sock[1], pj_ntohl(bound_addr.sin_addr.s_addr), + status=pj_sock_bind_in(sock[1], pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t)(next_rtp_port+1)); if (status != PJ_SUCCESS) { - pj_sock_close(sock[0]); + pj_sock_close(sock[0]); sock[0] = PJ_INVALID_SOCKET; - pj_sock_close(sock[1]); + pj_sock_close(sock[1]); sock[1] = PJ_INVALID_SOCKET; continue; } @@ -435,7 +554,7 @@ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, char ip_addr[32]; pj_str_t stun_srv; - pj_ansi_strcpy(ip_addr, + pj_ansi_strcpy(ip_addr, pj_inet_ntoa(pjsua_var.stun_srv.ipv4.sin_addr)); stun_srv = pj_str(ip_addr); @@ -449,23 +568,23 @@ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, } #if PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT - if (pj_ntohs(mapped_addr[1].sin_port) == + if (pj_ntohs(mapped_addr[1].sin_port) == pj_ntohs(mapped_addr[0].sin_port)+1) { /* Success! */ break; } - pj_sock_close(sock[0]); + pj_sock_close(sock[0]); sock[0] = PJ_INVALID_SOCKET; - pj_sock_close(sock[1]); + pj_sock_close(sock[1]); sock[1] = PJ_INVALID_SOCKET; #else - if (pj_ntohs(mapped_addr[1].sin_port) != + if (pj_ntohs(mapped_addr[1].sin_port) != pj_ntohs(mapped_addr[0].sin_port)+1) { - PJ_LOG(4,(THIS_FILE, + PJ_LOG(4,(THIS_FILE, "Note: STUN mapped RTCP port %d is not adjacent" " to RTP port %d", pj_ntohs(mapped_addr[1].sin_port), @@ -514,18 +633,18 @@ static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg, } if (sock[0] == PJ_INVALID_SOCKET) { - PJ_LOG(1,(THIS_FILE, + 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, + pj_memcpy(&skinfo->rtp_addr_name, &mapped_addr[0], sizeof(pj_sockaddr_in)); skinfo->rtcp_sock = sock[1]; - pj_memcpy(&skinfo->rtcp_addr_name, + pj_memcpy(&skinfo->rtcp_addr_name, &mapped_addr[1], sizeof(pj_sockaddr_in)); PJ_LOG(4,(THIS_FILE, "RTP socket reachable at %s", @@ -546,294 +665,106 @@ on_error: return status; } -/* Check if sound device is idle. */ -static void check_snd_dev_idle() -{ - unsigned call_cnt; - - /* Get the call count, we shouldn't close the sound device when there is - * any calls active. - */ - call_cnt = pjsua_call_get_count(); - - /* When this function is called from pjsua_media_channel_deinit() upon - * disconnecting call, actually the call count hasn't been updated/ - * decreased. So we put additional check here, if there is only one - * call and it's in DISCONNECTED state, there is actually no active - * call. - */ - if (call_cnt == 1) { - pjsua_call_id call_id; - pj_status_t status; - - status = pjsua_enum_calls(&call_id, &call_cnt); - if (status == PJ_SUCCESS && call_cnt > 0 && - !pjsua_call_is_active(call_id)) - { - call_cnt = 0; - } - } - - /* Activate sound device auto-close timer if sound device is idle. - * It is idle when there is no port connection in the bridge and - * there is no active call. - */ - 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 && - call_cnt == 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) +/* Create normal UDP media transports */ +static pj_status_t create_udp_media_transport(const pjsua_transport_config *cfg, + pjsua_call_media *call_med) { + pjmedia_sock_info skinfo; 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); - - /* Perform NAT detection */ - pjsua_detect_nat_type(); - - return PJ_SUCCESS; -} - - -/* - * Destroy pjsua media subsystem. - */ -pj_status_t pjsua_media_subsys_destroy(void) -{ - unsigned i; - - PJ_LOG(4,(THIS_FILE, "Shutting down media..")); - - 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; i<PJ_ARRAY_SIZE(pjsua_var.player); ++i) { - if (pjsua_var.player[i].port) { - pjmedia_port_destroy(pjsua_var.player[i].port); - pjsua_var.player[i].port = NULL; - } - } - - /* Destroy file recorders */ - for (i=0; i<PJ_ARRAY_SIZE(pjsua_var.recorder); ++i) { - if (pjsua_var.recorder[i].port) { - pjmedia_port_destroy(pjsua_var.recorder[i].port); - pjsua_var.recorder[i].port = NULL; - } + status = create_rtp_rtcp_sock(cfg, &skinfo); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to create RTP/RTCP socket", + status); + goto on_error; } - /* Close media transports */ - for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - if (pjsua_var.calls[i].med_tp_st != PJSUA_MED_TP_IDLE) { - pjsua_media_channel_deinit(i); - } - if (pjsua_var.calls[i].med_tp && pjsua_var.calls[i].med_tp_auto_del) { - pjmedia_transport_close(pjsua_var.calls[i].med_tp); - } - pjsua_var.calls[i].med_tp = NULL; + status = pjmedia_transport_udp_attach(pjsua_var.med_endpt, NULL, + &skinfo, 0, &call_med->tp); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to create media transport", + status); + goto on_error; } - /* Destroy media endpoint. */ - if (pjsua_var.med_endpt) { - - /* Shutdown all codecs: */ -# if PJMEDIA_HAS_SPEEX_CODEC - pjmedia_codec_speex_deinit(); -# endif /* PJMEDIA_HAS_SPEEX_CODEC */ - -# if PJMEDIA_HAS_GSM_CODEC - pjmedia_codec_gsm_deinit(); -# endif /* PJMEDIA_HAS_GSM_CODEC */ - -# if PJMEDIA_HAS_G711_CODEC - pjmedia_codec_g711_deinit(); -# endif /* PJMEDIA_HAS_G711_CODEC */ - -# if PJMEDIA_HAS_G722_CODEC - pjmedia_codec_g722_deinit(); -# endif /* PJMEDIA_HAS_G722_CODEC */ - -# if PJMEDIA_HAS_INTEL_IPP - pjmedia_codec_ipp_deinit(); -# endif /* PJMEDIA_HAS_INTEL_IPP */ + pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING, + pjsua_var.media_cfg.tx_drop_pct); -# if PJMEDIA_HAS_PASSTHROUGH_CODECS - pjmedia_codec_passthrough_deinit(); -# endif /* PJMEDIA_HAS_PASSTHROUGH_CODECS */ + pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING, + pjsua_var.media_cfg.rx_drop_pct); -# if PJMEDIA_HAS_G7221_CODEC - pjmedia_codec_g7221_deinit(); -# endif /* PJMEDIA_HAS_G7221_CODEC */ - -# if PJMEDIA_HAS_L16_CODEC - pjmedia_codec_l16_deinit(); -# endif /* PJMEDIA_HAS_L16_CODEC */ - - pjmedia_endpt_destroy(pjsua_var.med_endpt); - pjsua_var.med_endpt = NULL; - - /* Deinitialize sound subsystem */ - // Not necessary, as pjmedia_snd_deinit() should have been called - // in pjmedia_endpt_destroy(). - //pjmedia_snd_deinit(); - } + return PJ_SUCCESS; - /* Reset RTP port */ - next_rtp_port = 0; +on_error: + if (call_med->tp) + pjmedia_transport_close(call_med->tp); - return PJ_SUCCESS; + return status; } - +#if DISABLED_FOR_TICKET_1185 /* Create normal UDP media transports */ static pj_status_t create_udp_media_transports(pjsua_transport_config *cfg) { unsigned i; - pjmedia_sock_info skinfo; pj_status_t status; - /* Create each media transport */ - for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { + for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) { + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; - status = create_rtp_rtcp_sock(cfg, &skinfo); - if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Unable to create RTP/RTCP socket", - status); - goto on_error; - } + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; - status = pjmedia_transport_udp_attach(pjsua_var.med_endpt, NULL, - &skinfo, 0, - &pjsua_var.calls[i].med_tp); - if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Unable to create media transport", - status); - goto on_error; + status = create_udp_media_transport(cfg, &call_med->tp); + if (status != PJ_SUCCESS) + goto on_error; } - - pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp, - PJMEDIA_DIR_ENCODING, - pjsua_var.media_cfg.tx_drop_pct); - - pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp, - PJMEDIA_DIR_DECODING, - pjsua_var.media_cfg.rx_drop_pct); - } return PJ_SUCCESS; on_error: - for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - if (pjsua_var.calls[i].med_tp != NULL) { - pjmedia_transport_close(pjsua_var.calls[i].med_tp); - pjsua_var.calls[i].med_tp = NULL; + for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) { + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; + + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + + if (call_med->tp) { + pjmedia_transport_close(call_med->tp); + call_med->tp = NULL; + } } } - return status; } - +#endif /* This callback is called when ICE negotiation completes */ static void on_ice_complete(pjmedia_transport *tp, pj_ice_strans_op op, pj_status_t result) { - unsigned id; - pj_bool_t found = PJ_FALSE; - - /* Find call which has this media transport */ + pjsua_call_media *call_med = (pjsua_call_media*)tp->user_data; - PJSUA_LOCK(); - - for (id=0; id<pjsua_var.ua_cfg.max_calls; ++id) { - if (pjsua_var.calls[id].med_tp == tp || - pjsua_var.calls[id].med_orig == tp) - { - found = PJ_TRUE; - break; - } - } - - PJSUA_UNLOCK(); - - if (!found) + if (!call_med) return; switch (op) { case PJ_ICE_STRANS_OP_INIT: - pjsua_var.calls[id].med_tp_ready = result; + call_med->tp_ready = result; break; case PJ_ICE_STRANS_OP_NEGOTIATION: if (result != PJ_SUCCESS) { - pjsua_var.calls[id].media_st = PJSUA_CALL_MEDIA_ERROR; - pjsua_var.calls[id].media_dir = PJMEDIA_DIR_NONE; + call_med->state = PJSUA_CALL_MEDIA_ERROR; + call_med->dir = PJMEDIA_DIR_NONE; - if (pjsua_var.ua_cfg.cb.on_call_media_state) { - pjsua_var.ua_cfg.cb.on_call_media_state(id); + if (call_med->call && pjsua_var.ua_cfg.cb.on_call_media_state) { + pjsua_var.ua_cfg.cb.on_call_media_state(call_med->call->index); } - } else { + } else if (call_med->call) { /* Send UPDATE if default transport address is different than * what was advertised (ticket #881) */ @@ -853,36 +784,38 @@ static void on_ice_complete(pjmedia_transport *tp, if (ii && ii->role==PJ_ICE_SESS_ROLE_CONTROLLING && pj_sockaddr_cmp(&tpinfo.sock_info.rtp_addr_name, - &pjsua_var.calls[id].med_rtp_addr)) + &call_med->rtp_addr)) { pj_bool_t use_update; const pj_str_t STR_UPDATE = { "UPDATE", 6 }; pjsip_dialog_cap_status support_update; pjsip_dialog *dlg; - dlg = pjsua_var.calls[id].inv->dlg; + dlg = call_med->call->inv->dlg; support_update = pjsip_dlg_remote_has_cap(dlg, PJSIP_H_ALLOW, NULL, &STR_UPDATE); use_update = (support_update == PJSIP_DIALOG_CAP_SUPPORTED); PJ_LOG(4,(THIS_FILE, "ICE default transport address has changed for " - "call %d, sending %s", id, + "call %d, sending %s", call_med->call->index, (use_update ? "UPDATE" : "re-INVITE"))); if (use_update) - pjsua_call_update(id, 0, NULL); + pjsua_call_update(call_med->call->index, 0, NULL); else - pjsua_call_reinvite(id, 0, NULL); + pjsua_call_reinvite(call_med->call->index, 0, NULL); } } break; case PJ_ICE_STRANS_OP_KEEP_ALIVE: if (result != PJ_SUCCESS) { PJ_PERROR(4,(THIS_FILE, result, - "ICE keep alive failure for transport %d", id)); + "ICE keep alive failure for transport %d:%d", + call_med->call->index, call_med->idx)); } if (pjsua_var.ua_cfg.cb.on_ice_transport_error) { + pjsua_call_id id = call_med->call->index; (*pjsua_var.ua_cfg.cb.on_ice_transport_error)(id, op, result, NULL); } @@ -918,11 +851,15 @@ static pj_status_t parse_host_port(const pj_str_t *host_port, } /* Create ICE media transports (when ice is enabled) */ -static pj_status_t create_ice_media_transports(pjsua_transport_config *cfg) +static pj_status_t create_ice_media_transport( + const pjsua_transport_config *cfg, + pjsua_call_media *call_med) { char stunip[PJ_INET6_ADDRSTRLEN]; pj_ice_strans_cfg ice_cfg; - unsigned i; + pjmedia_ice_cb ice_cb; + char name[32]; + unsigned comp_cnt; pj_status_t status; /* Make sure STUN server resolution has completed */ @@ -979,68 +916,97 @@ static pj_status_t create_ice_media_transports(pjsua_transport_config *cfg) sizeof(cfg->qos_params)); } - /* Create each media transport */ - for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - pjmedia_ice_cb ice_cb; - char name[32]; - unsigned comp_cnt; - - pj_bzero(&ice_cb, sizeof(pjmedia_ice_cb)); - ice_cb.on_ice_complete = &on_ice_complete; - pj_ansi_snprintf(name, sizeof(name), "icetp%02d", i); - pjsua_var.calls[i].med_tp_ready = PJ_EPENDING; - - comp_cnt = 1; - if (PJMEDIA_ADVERTISE_RTCP && !pjsua_var.media_cfg.ice_no_rtcp) - ++comp_cnt; - - status = pjmedia_ice_create(pjsua_var.med_endpt, name, comp_cnt, - &ice_cfg, &ice_cb, - &pjsua_var.calls[i].med_tp); - if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Unable to create ICE media transport", - status); - goto on_error; - } + pj_bzero(&ice_cb, sizeof(pjmedia_ice_cb)); + ice_cb.on_ice_complete = &on_ice_complete; + pj_ansi_snprintf(name, sizeof(name), "icetp%02d", call_med->idx); + call_med->tp_ready = PJ_EPENDING; - /* Wait until transport is initialized, or time out */ - PJSUA_UNLOCK(); - while (pjsua_var.calls[i].med_tp_ready == PJ_EPENDING) { - pjsua_handle_events(100); - } - PJSUA_LOCK(); - if (pjsua_var.calls[i].med_tp_ready != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Error initializing ICE media transport", - pjsua_var.calls[i].med_tp_ready); - status = pjsua_var.calls[i].med_tp_ready; - goto on_error; - } + comp_cnt = 1; + if (PJMEDIA_ADVERTISE_RTCP && !pjsua_var.media_cfg.ice_no_rtcp) + ++comp_cnt; - pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp, - PJMEDIA_DIR_ENCODING, - pjsua_var.media_cfg.tx_drop_pct); + status = pjmedia_ice_create3(pjsua_var.med_endpt, name, comp_cnt, + &ice_cfg, &ice_cb, 0, call_med, + &call_med->tp); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to create ICE media transport", + status); + goto on_error; + } - pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp, - PJMEDIA_DIR_DECODING, - pjsua_var.media_cfg.rx_drop_pct); + /* Wait until transport is initialized, or time out */ + PJSUA_UNLOCK(); + while (call_med->tp_ready == PJ_EPENDING) { + pjsua_handle_events(100); + } + PJSUA_LOCK(); + if (call_med->tp_ready != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Error initializing ICE media transport", + call_med->tp_ready); + status = call_med->tp_ready; + goto on_error; } + pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING, + pjsua_var.media_cfg.tx_drop_pct); + + pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING, + pjsua_var.media_cfg.rx_drop_pct); + return PJ_SUCCESS; on_error: - for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - if (pjsua_var.calls[i].med_tp != NULL) { - pjmedia_transport_close(pjsua_var.calls[i].med_tp); - pjsua_var.calls[i].med_tp = NULL; - } + if (call_med->tp != NULL) { + pjmedia_transport_close(call_med->tp); + call_med->tp = NULL; } return status; } +#if DISABLED_FOR_TICKET_1185 +/* Create ICE media transports (when ice is enabled) */ +static pj_status_t create_ice_media_transports(pjsua_transport_config *cfg) +{ + unsigned i; + pj_status_t status; + + for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) { + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; + + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + + status = create_ice_media_transport(cfg, call_med); + if (status != PJ_SUCCESS) + goto on_error; + } + } + + return PJ_SUCCESS; + +on_error: + for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) { + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; + + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + + if (call_med->tp) { + pjmedia_transport_close(call_med->tp); + call_med->tp = NULL; + } + } + } + return status; +} +#endif +#if DISABLED_FOR_TICKET_1185 /* - * Create UDP media transports for all the calls. This function creates + * Create media transports for all the calls. This function creates * one UDP media transport for each call. */ PJ_DEF(pj_status_t) pjsua_media_transports_create( @@ -1058,12 +1024,17 @@ PJ_DEF(pj_status_t) pjsua_media_transports_create( /* Delete existing media transports */ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - if (pjsua_var.calls[i].med_tp != NULL && - pjsua_var.calls[i].med_tp_auto_del) - { - pjmedia_transport_close(pjsua_var.calls[i].med_tp); - pjsua_var.calls[i].med_tp = NULL; - pjsua_var.calls[i].med_orig = NULL; + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; + + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + + if (call_med->tp && call_med->tp_auto_del) { + pjmedia_transport_close(call_med->tp); + call_med->tp = NULL; + call_med->tp_orig = NULL; + } } } @@ -1079,7 +1050,14 @@ PJ_DEF(pj_status_t) pjsua_media_transports_create( /* Set media transport auto_delete to True */ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - pjsua_var.calls[i].med_tp_auto_del = PJ_TRUE; + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; + + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + + call_med->tp_auto_del = PJ_TRUE; + } } PJSUA_UNLOCK(); @@ -1100,184 +1078,421 @@ PJ_DEF(pj_status_t) pjsua_media_transports_attach(pjsua_media_transport tp[], /* Assign the media transports */ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) { - if (pjsua_var.calls[i].med_tp != NULL && - pjsua_var.calls[i].med_tp_auto_del) - { - pjmedia_transport_close(pjsua_var.calls[i].med_tp); + pjsua_call *call = &pjsua_var.calls[i]; + unsigned strm_idx; + + for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) { + pjsua_call_media *call_med = &call->media[strm_idx]; + + if (call_med->tp && call_med->tp_auto_del) { + pjmedia_transport_close(call_med->tp); + call_med->tp = NULL; + call_med->tp_orig = NULL; + } } - pjsua_var.calls[i].med_tp = tp[i].transport; - pjsua_var.calls[i].med_tp_auto_del = auto_delete; + PJ_TODO(remove_pjsua_media_transports_attach); + + call->media[0].tp = tp[i].transport; + call->media[0].tp_auto_del = auto_delete; } return PJ_SUCCESS; } +#endif - -static int find_audio_index(const pjmedia_sdp_session *sdp, - pj_bool_t prefer_srtp) +/* Go through the list of media in the SDP, find acceptable media, and + * sort them based on the "quality" of the media, and store the indexes + * in the specified array. Media with the best quality will be listed + * first in the array. The quality factors considered currently is + * encryption. + */ +static void sort_media(const pjmedia_sdp_session *sdp, + const pj_str_t *type, + pjmedia_srtp_use use_srtp, + pj_uint8_t midx[], + unsigned *p_count) { unsigned i; - int audio_idx = -1; + unsigned count = 0; + int score[PJSUA_MAX_CALL_MEDIA]; - for (i=0; i<sdp->media_count; ++i) { - const pjmedia_sdp_media *m = sdp->media[i]; + pj_assert(*p_count >= PJSUA_MAX_CALL_MEDIA); - /* Skip if media is not audio */ - if (pj_stricmp2(&m->desc.media, "audio") != 0) - continue; + *p_count = 0; + for (i=0; i<PJSUA_MAX_CALL_MEDIA; ++i) + score[i] = 1; - /* Skip if media is disabled */ - if (m->desc.port == 0) - continue; + /* Score each media */ + for (i=0; i<sdp->media_count && count<PJSUA_MAX_CALL_MEDIA; ++i) { + const pjmedia_sdp_media *m = sdp->media[i]; + const pjmedia_sdp_conn *c; - /* Skip if transport is not supported */ - if (pj_stricmp2(&m->desc.transport, "RTP/AVP") != 0 && - pj_stricmp2(&m->desc.transport, "RTP/SAVP") != 0) - { + /* Skip different media */ + if (pj_stricmp(&m->desc.media, type) != 0) { + score[count++] = -22000; continue; } - if (audio_idx == -1) { - audio_idx = i; - } else { - /* We've found multiple candidates. This could happen - * e.g. when remote is offering both RTP/SAVP and RTP/AVP, - * or when remote for some reason offers two audio. - */ + c = m->conn? m->conn : sdp->conn; - if (prefer_srtp && - pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0) - { - /* Prefer RTP/SAVP when our media transport is SRTP */ - audio_idx = i; + /* Supported transports */ + if (pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0) { + switch (use_srtp) { + case PJMEDIA_SRTP_MANDATORY: + case PJMEDIA_SRTP_OPTIONAL: + ++score[i]; + break; + case PJMEDIA_SRTP_DISABLED: + --score[i]; break; - } else if (!prefer_srtp && - pj_stricmp2(&m->desc.transport, "RTP/AVP")==0) - { - /* Prefer RTP/AVP when our media transport is NOT SRTP */ - audio_idx = i; } + } else if (pj_stricmp2(&m->desc.transport, "RTP/AVP")==0) { + switch (use_srtp) { + case PJMEDIA_SRTP_MANDATORY: + --score[i]; + break; + case PJMEDIA_SRTP_OPTIONAL: + /* No change in score */ + break; + case PJMEDIA_SRTP_DISABLED: + ++score[i]; + break; + } + } else { + score[i] -= 10; } + + /* Is media disabled? */ + if (m->desc.port == 0) + score[i] -= 10; + + /* Is media inactive? */ + if (pjmedia_sdp_media_find_attr2(m, "inactive", NULL) || + pj_strcmp2(&c->addr, "0.0.0.0") == 0) + { + //score[i] -= 10; + score[i] -= 1; + } + + ++count; } - return audio_idx; + /* Created sorted list based on quality */ + for (i=0; i<count; ++i) { + unsigned j; + int best = 0; + + for (j=1; j<count; ++j) { + if (score[j] > score[best]) + best = j; + } + /* Don't put media with negative score, that media is unacceptable + * for us. + */ + if (score[best] >= 0) { + midx[*p_count] = (pj_uint8_t)best; + (*p_count)++; + } + + score[best] = -22000; + + } } +/* Callback to receive media events */ +static pj_status_t call_media_on_event(pjmedia_event_subscription *esub, + pjmedia_event *event) +{ + pjsua_call_media *call_med = (pjsua_call_media*)esub->user_data; + pjsua_call *call = call_med->call; -pj_status_t pjsua_media_channel_init(pjsua_call_id call_id, - pjsip_role_e role, - int security_level, - pj_pool_t *tmp_pool, - const pjmedia_sdp_session *rem_sdp, - int *sip_err_code) + if (pjsua_var.ua_cfg.cb.on_call_media_event && call) { + ++event->proc_cnt; + (*pjsua_var.ua_cfg.cb.on_call_media_event)(call->index, + call_med->idx, event); + } + + return PJ_SUCCESS; +} + +/* Initialize the media line */ +pj_status_t pjsua_call_media_init(pjsua_call_media *call_med, + pjmedia_type type, + const pjsua_transport_config *tcfg, + int security_level, + int *sip_err_code) { - pjsua_call *call = &pjsua_var.calls[call_id]; + pjsua_acc *acc = &pjsua_var.acc[call_med->call->acc_id]; pj_status_t status; -#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) - pjsua_acc *acc = &pjsua_var.acc[call->acc_id]; - pjmedia_srtp_setting srtp_opt; - pjmedia_transport *srtp = NULL; -#endif + /* + * Note: this function may be called when the media already exists + * (e.g. in reinvites, updates, etc.) + */ + call_med->type = type; - PJ_UNUSED_ARG(role); + /* Create the media transport for initial call. This is blocking for now */ + if (call_med->tp == NULL) { + if (pjsua_var.media_cfg.enable_ice) { + status = create_ice_media_transport(tcfg, call_med); + } else { + status = create_udp_media_transport(tcfg, call_med); + } - /* 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 (status != PJ_SUCCESS) { + PJ_PERROR(1,(THIS_FILE, status, "Error creating media transport")); + return status; + } + + call_med->tp_st = PJSUA_MED_TP_IDLE; + + /* While in initial call, set default video devices */ + if (type == PJMEDIA_TYPE_VIDEO) { + call_med->strm.v.rdr_dev = acc->cfg.vid_rend_dev; + call_med->strm.v.cap_dev = acc->cfg.vid_cap_dev; + if (call_med->strm.v.rdr_dev == PJMEDIA_VID_DEFAULT_RENDER_DEV) { + pjmedia_vid_dev_info info; + pjmedia_vid_dev_get_info(call_med->strm.v.rdr_dev, &info); + call_med->strm.v.rdr_dev = info.id; + } + if (call_med->strm.v.cap_dev == PJMEDIA_VID_DEFAULT_CAPTURE_DEV) { + pjmedia_vid_dev_info info; + pjmedia_vid_dev_get_info(call_med->strm.v.cap_dev, &info); + call_med->strm.v.cap_dev = info.id; + } + } + } else if (call_med->tp_st == PJSUA_MED_TP_DISABLED) { + /* Media is being reenabled. */ + call_med->tp_st = PJSUA_MED_TP_INIT; } #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) - /* This function may be called when SRTP transport already exists + /* 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) { + if (!call_med->tp_orig || call_med->tp == call_med->tp_orig) { + pjmedia_srtp_setting srtp_opt; + pjmedia_transport *srtp = NULL; /* 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; + status = PJSIP_ESESSIONINSECURE; + goto on_error; } } /* 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 + srtp_opt.close_member_tp = PJ_TRUE; + /* 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; + if (call_med->rem_srtp_use > acc->cfg.use_srtp) + srtp_opt.use = call_med->rem_srtp_use; else srtp_opt.use = acc->cfg.use_srtp; - status = pjmedia_transport_srtp_create(pjsua_var.med_endpt, - call->med_tp, + 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; + goto on_error; } /* Set SRTP as current media transport */ - call->med_orig = call->med_tp; - call->med_tp = srtp; + call_med->tp_orig = call_med->tp; + call_med->tp = srtp; } #else - call->med_orig = call->med_tp; + call->tp_orig = call->tp; PJ_UNUSED_ARG(security_level); #endif - /* Find out which media line in SDP that we support. If we are offerer, - * audio will be initialized at index 0 in SDP. + pjmedia_event_subscription_init(&call_med->esub_rend, &call_media_on_event, + call_med); + pjmedia_event_subscription_init(&call_med->esub_cap, &call_media_on_event, + call_med); + + return PJ_SUCCESS; + +on_error: + if (call_med->tp) { + pjmedia_transport_close(call_med->tp); + call_med->tp = NULL; + } + return status; +} + +pj_status_t pjsua_media_channel_init(pjsua_call_id call_id, + pjsip_role_e role, + int security_level, + pj_pool_t *tmp_pool, + const pjmedia_sdp_session *rem_sdp, + int *sip_err_code) +{ + const pj_str_t STR_AUDIO = { "audio", 5 }; + const pj_str_t STR_VIDEO = { "video", 5 }; + pjsua_call *call = &pjsua_var.calls[call_id]; + pjsua_acc *acc = &pjsua_var.acc[call->acc_id]; + pj_uint8_t maudidx[PJSUA_MAX_CALL_MEDIA]; + unsigned maudcnt = PJ_ARRAY_SIZE(maudidx); + pj_uint8_t mvididx[PJSUA_MAX_CALL_MEDIA]; + unsigned mvidcnt = PJ_ARRAY_SIZE(mvididx); + pjmedia_type media_types[PJSUA_MAX_CALL_MEDIA]; + unsigned mi; + pj_status_t status; + + PJ_UNUSED_ARG(role); + + /* + * Note: this function may be called when the media already exists + * (e.g. in reinvites, updates, etc). */ - if (rem_sdp == NULL) { - call->audio_idx = 0; - } - /* Otherwise find out the candidate audio media line in SDP */ - else { - pj_bool_t srtp_active; -#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) - srtp_active = acc->cfg.use_srtp; -#else - srtp_active = PJ_FALSE; + if (pjsua_get_state() != PJSUA_STATE_RUNNING) + return PJ_EBUSY; + +#if DISABLED_FOR_TICKET_1185 + /* Return error if media transport has not been created yet + * (e.g. application is starting) + */ + for (i=0; i<call->med_cnt; ++i) { + if (call->media[i].tp == NULL) { + return PJ_EBUSY; + } + } #endif - /* Media count must have been checked */ - pj_assert(rem_sdp->media_count != 0); + if (rem_sdp) { + sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp, + maudidx, &maudcnt); + if (maudcnt > acc->cfg.max_audio_cnt) + maudcnt = acc->cfg.max_audio_cnt; + + if (maudcnt==0) { + /* Expecting audio in the offer */ + 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); + } + + sort_media(rem_sdp, &STR_VIDEO, acc->cfg.use_srtp, + mvididx, &mvidcnt); + if (mvidcnt > acc->cfg.max_video_cnt) + mvidcnt = acc->cfg.max_video_cnt; + + /* Update media count only when remote add any media, this media count + * must never decrease. + */ + if (call->med_cnt < rem_sdp->media_count) + call->med_cnt = PJ_MIN(rem_sdp->media_count, PJSUA_MAX_CALL_MEDIA); - call->audio_idx = find_audio_index(rem_sdp, srtp_active); + } else { + maudcnt = acc->cfg.max_audio_cnt; + for (mi=0; mi<maudcnt; ++mi) { + maudidx[mi] = (pj_uint8_t)mi; + media_types[mi] = PJMEDIA_TYPE_AUDIO; + } + mvidcnt = acc->cfg.max_video_cnt; + for (mi=0; mi<mvidcnt; ++mi) { + media_types[maudcnt + mi] = PJMEDIA_TYPE_VIDEO; + } + + call->med_cnt = maudcnt + mvidcnt; } - /* Reject offer if we couldn't find a good m=audio line in offer */ - if (call->audio_idx < 0) { + if (call->med_cnt == 0) { + /* Expecting at least one media */ 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", + /* Initialize each media line */ + for (mi=0; mi < call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; + pj_bool_t enabled = PJ_FALSE; + pjmedia_type media_type = PJMEDIA_TYPE_NONE; + + if (rem_sdp) { + if (mi >= rem_sdp->media_count) { + /* Media has been removed in remote re-offer */ + media_type = call_med->type; + } else if (!pj_stricmp(&rem_sdp->media[mi]->desc.media, &STR_AUDIO)) { + media_type = PJMEDIA_TYPE_AUDIO; + if (pj_memchr(maudidx, mi, maudcnt * sizeof(maudidx[0]))) { + enabled = PJ_TRUE; + } + } + else if (!pj_stricmp(&rem_sdp->media[mi]->desc.media, &STR_VIDEO)) { + media_type = PJMEDIA_TYPE_VIDEO; + if (pj_memchr(mvididx, mi, mvidcnt * sizeof(mvididx[0]))) { + enabled = PJ_TRUE; + } + } + + } else { + enabled = PJ_TRUE; + media_type = media_types[mi]; + } + + if (enabled) { + status = pjsua_call_media_init(call_med, media_type, + &acc->cfg.rtp_cfg, + security_level, sip_err_code); + if (status != PJ_SUCCESS) { + pjsua_media_channel_deinit(call_id); + return status; + } + } else { + /* By convention, the media is disabled if transport is NULL + * or transport state is PJSUA_MED_TP_DISABLED. + */ + if (call_med->tp) { + // Don't close transport here, as SDP negotiation has not been + // done and stream may be still active. + //pjmedia_transport_close(call_med->tp); + //call_med->tp = NULL; + pj_assert(call_med->tp_st == PJSUA_MED_TP_INIT || + call_med->tp_st == PJSUA_MED_TP_RUNNING); + call_med->tp_st = PJSUA_MED_TP_DISABLED; + } + + /* Put media type just for info */ + call_med->type = media_type; + } + } + + call->audio_idx = maudidx[0]; + + PJ_LOG(4,(THIS_FILE, "Media index %d selected for audio 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; + /* Tell the media transport of a new offer/answer session */ + for (mi=0; mi < call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; + + /* Note: tp may be NULL if this media line is disabled */ + if (call_med->tp && call_med->tp_st == PJSUA_MED_TP_IDLE) { + status = pjmedia_transport_media_create(call_med->tp, + tmp_pool, 0, + rem_sdp, mi); + 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; + } } - call->med_tp_st = PJSUA_MED_TP_INIT; return PJ_SUCCESS; } @@ -1285,106 +1500,183 @@ 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) + int *sip_err_code) { - enum { MAX_MEDIA = 1 }; + enum { MAX_MEDIA = PJSUA_MAX_CALL_MEDIA }; pjmedia_sdp_session *sdp; - pjmedia_transport_info tpinfo; + pj_sockaddr origin; pjsua_call *call = &pjsua_var.calls[call_id]; pjmedia_sdp_neg_state sdp_neg_state = PJMEDIA_SDP_NEG_STATE_NULL; + unsigned mi; pj_status_t status; - /* Return error if media transport has not been created yet - * (e.g. application is starting) - */ - if (call->med_tp == NULL) { + if (pjsua_get_state() != PJSUA_STATE_RUNNING) return PJ_EBUSY; - } - - if (rem_sdp && rem_sdp->media_count != 0) { - pj_bool_t srtp_active; -#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) - srtp_active = pjsua_var.acc[call->acc_id].cfg.use_srtp; -#else - srtp_active = PJ_FALSE; -#endif + if (rem_sdp) { + /* If this is a re-offer, let's re-initialize media as remote may + * add or remove media + */ + if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) { + status = pjsua_media_channel_init(call_id, PJSIP_ROLE_UAS, + call->secure_level, pool, + rem_sdp, sip_err_code); + if (status != PJ_SUCCESS) + return status; + } - call->audio_idx = find_audio_index(rem_sdp, srtp_active); - if (call->audio_idx == -1) { - /* No audio in the offer. We can't accept this */ - PJ_LOG(4,(THIS_FILE, - "Unable to accept SDP offer without audio for call %d", - call_id)); - return PJMEDIA_SDP_EINMEDIA; +#if 0 + pjsua_acc *acc = &pjsua_var.acc[call->acc_id]; + pj_uint8_t maudidx[PJSUA_MAX_CALL_MEDIA]; + unsigned maudcnt = PJ_ARRAY_SIZE(maudidx); + + sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp, + maudidx, &maudcnt); + + if (maudcnt==0) { + /* Expecting audio in the offer */ + 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); } - } - /* Media index must have been determined before */ - pj_assert(call->audio_idx != -1); + call->audio_idx = maudidx[0]; +#endif + } else { + /* Audio is first in our offer, by convention */ + // The audio_idx should not be changed here, as this function may be + // called in generating re-offer and the current active audio index + // can be anywhere. + //call->audio_idx = 0; + } +#if 0 + // Since r3512, old-style hold should have got transport, created by + // pjsua_media_channel_init() in initial offer/answer or remote reoffer. /* Create media if it's not created. This could happen when call is - * currently on-hold + * currently on-hold (with the old style hold) */ - if (call->med_tp_st == PJSUA_MED_TP_IDLE) { + if (call->media[call->audio_idx].tp == NULL) { 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); + pool, rem_sdp, sip_err_code); if (status != PJ_SUCCESS) return status; } +#endif /* Get SDP negotiator state */ if (call->inv && call->inv->neg) sdp_neg_state = pjmedia_sdp_neg_get_state(call->inv->neg); - /* Get media socket info */ - pjmedia_transport_info_init(&tpinfo); - pjmedia_transport_get_info(call->med_tp, &tpinfo); + /* Get one address to use in the origin field */ + pj_bzero(&origin, sizeof(origin)); + for (mi=0; mi<call->med_cnt; ++mi) { + pjmedia_transport_info 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 (call->media[mi].tp == NULL) + continue; + + pjmedia_transport_info_init(&tpinfo); + pjmedia_transport_get_info(call->media[mi].tp, &tpinfo); + pj_sockaddr_cp(&origin, &tpinfo.sock_info.rtp_addr_name); + break; } - /* If we're answering or updating the session with a new offer, - * 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 (call->audio_idx != 0 && - (rem_sdp || sdp_neg_state==PJMEDIA_SDP_NEG_STATE_DONE)) - { - unsigned i; - const pjmedia_sdp_session *ref_sdp = rem_sdp; + /* Create the base (blank) SDP */ + status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, pool, NULL, + &origin, &sdp); + if (status != PJ_SUCCESS) + return status; + + /* Process each media line */ + for (mi=0; mi<call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; + pjmedia_sdp_media *m = NULL; + pjmedia_transport_info tpinfo; - if (!ref_sdp) { - /* We are updating session with a new offer */ - status = pjmedia_sdp_neg_get_active_local(call->inv->neg, - &ref_sdp); - pj_assert(status == PJ_SUCCESS); + if (rem_sdp && mi >= rem_sdp->media_count) { + /* Remote might have removed some media lines. */ + break; } - for (i=0; i<ref_sdp->media_count; ++i) { - const pjmedia_sdp_media *ref_m = ref_sdp->media[i]; - pjmedia_sdp_media *m; + if (call_med->tp == NULL || call_med->tp_st == PJSUA_MED_TP_DISABLED) + { + /* + * This media is disabled. Just create a valid SDP with zero + * port. + */ + m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media); + m->desc.transport = pj_str("RTP/AVP"); + m->desc.fmt_count = 1; + m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn); + m->conn->net_type = pj_str("IN"); + m->conn->addr_type = pj_str("IP4"); + m->conn->addr = pj_str("127.0.0.1"); + + switch (call_med->type) { + case PJMEDIA_TYPE_AUDIO: + m->desc.media = pj_str("audio"); + m->desc.fmt[0] = pj_str("0"); + break; + case PJMEDIA_TYPE_VIDEO: + m->desc.media = pj_str("video"); + m->desc.fmt[0] = pj_str("31"); + break; + default: + if (rem_sdp && mi < rem_sdp->media_count) { + pj_strdup(pool, &m->desc.media, + &rem_sdp->media[mi]->desc.media); + pj_strdup(pool, &m->desc.fmt[0], + &rem_sdp->media[mi]->desc.fmt[0]); + } else { + pj_assert(!"Invalid call_med media type"); + return PJ_EBUG; + } + } + + sdp->media[sdp->media_count++] = m; + continue; + } - if ((int)i == call->audio_idx) - continue; + /* Get transport address info */ + pjmedia_transport_info_init(&tpinfo); + pjmedia_transport_get_info(call_med->tp, &tpinfo); - m = pjmedia_sdp_media_clone_deactivate(pool, ref_m); - 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; - } + /* Ask pjmedia endpoint to create SDP media line */ + switch (call_med->type) { + case PJMEDIA_TYPE_AUDIO: + status = pjmedia_endpt_create_audio_sdp(pjsua_var.med_endpt, pool, + &tpinfo.sock_info, 0, &m); + break; + case PJMEDIA_TYPE_VIDEO: + status = pjmedia_endpt_create_video_sdp(pjsua_var.med_endpt, pool, + &tpinfo.sock_info, 0, &m); + break; + default: + pj_assert(!"Invalid call_med media type"); + return PJ_EBUG; + } + + if (status != PJ_SUCCESS) + return status; + + sdp->media[sdp->media_count++] = m; + + /* Give to transport */ + status = pjmedia_transport_encode_sdp(call_med->tp, pool, + sdp, rem_sdp, mi); + if (status != PJ_SUCCESS) { + if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE; + return status; + } + + /* Copy c= line of the first media to session level, + * if there's none. + */ + if (sdp->conn == NULL) { + sdp->conn = pjmedia_sdp_conn_clone(pool, m->conn); } } @@ -1412,15 +1704,8 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id, } - /* 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; - } -#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) +#if DISABLED_FOR_TICKET_1185 && defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0) /* Check if SRTP is in optional mode and configured to use duplicated * media, i.e: secured and unsecured version, in the SDP offer. */ @@ -1433,13 +1718,13 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id, for (i = 0; i < sdp->media_count; ++i) { pjmedia_sdp_media *m = sdp->media[i]; - /* Check if this media is unsecured but has SDP "crypto" + /* Check if this media is unsecured but has SDP "crypto" * attribute. */ if (pj_stricmp2(&m->desc.transport, "RTP/AVP") == 0 && pjmedia_sdp_media_find_attr2(m, "crypto", NULL) != NULL) { - if (i == (unsigned)call->audio_idx && + if (i == (unsigned)call->audio_idx && sdp_neg_state == PJMEDIA_SDP_NEG_STATE_DONE) { /* This is a session update, and peer has chosen the @@ -1462,7 +1747,7 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id, /* Insert the new media before the unsecured media */ if (sdp->media_count < PJMEDIA_MAX_SDP_MEDIA) { - pj_array_insert(sdp->media, sizeof(new_m), + pj_array_insert(sdp->media, sizeof(new_m), sdp->media_count, i, &new_m); ++sdp->media_count; ++i; @@ -1473,10 +1758,6 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id, } #endif - /* Update currently advertised RTP source address */ - pj_memcpy(&call->med_rtp_addr, &tpinfo.sock_info.rtp_addr_name, - sizeof(pj_sockaddr)); - *p_sdp = sdp; return PJ_SUCCESS; } @@ -1485,61 +1766,83 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id, static void stop_media_session(pjsua_call_id call_id) { pjsua_call *call = &pjsua_var.calls[call_id]; + unsigned mi; - if (call->conf_slot != PJSUA_INVALID_ID) { - if (pjsua_var.mconf) { - pjsua_conf_remove_port(call->conf_slot); - } - call->conf_slot = PJSUA_INVALID_ID; - } + for (mi=0; mi<call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; - if (call->session) { - pjmedia_rtcp_stat stat; + if (call_med->type == PJMEDIA_TYPE_AUDIO) { + pjmedia_stream *strm = call_med->strm.a.stream; + pjmedia_rtcp_stat stat; - if ((call->media_dir & PJMEDIA_DIR_ENCODING) && - (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 (strm) { + if (call_med->strm.a.conf_slot != PJSUA_INVALID_ID) { + if (pjsua_var.mconf) { + pjsua_conf_remove_port(call_med->strm.a.conf_slot); + } + call_med->strm.a.conf_slot = PJSUA_INVALID_ID; + } - if (pjsua_var.ua_cfg.cb.on_stream_destroyed) { - pjsua_var.ua_cfg.cb.on_stream_destroyed(call_id, call->session, 0); - } + if ((call_med->dir & PJMEDIA_DIR_ENCODING) && + (pjmedia_stream_get_stat(strm, &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_med->rtp_tx_seq_ts_set = 1 | (1 << 1); + call_med->rtp_tx_seq = stat.rtp_tx_last_seq; + call_med->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, strm, mi); + } - pjmedia_session_destroy(call->session); - call->session = NULL; + pjmedia_stream_destroy(strm); + call_med->strm.a.stream = NULL; + } + } - PJ_LOG(4,(THIS_FILE, "Media session for call %d is destroyed", - call_id)); +#if PJMEDIA_HAS_VIDEO + else if (call_med->type == PJMEDIA_TYPE_VIDEO) { + stop_video_stream(call_med); + } +#endif + PJ_LOG(4,(THIS_FILE, "Media session call%02d:%d is destroyed", + call_id, mi)); + call_med->state = PJSUA_CALL_MEDIA_NONE; } - - 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]; + unsigned mi; 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; - } + for (mi=0; mi<call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; - 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; + 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->tp_orig && call_med->tp && + // call_med->tp != call_med->tp_orig) + //{ + // pjmedia_transport_close(call_med->tp); + // call_med->tp = call_med->tp_orig; + //} + if (call_med->tp) { + pjmedia_transport_close(call_med->tp); + call_med->tp = call_med->tp_orig = NULL; + } } check_snd_dev_idle(); @@ -1569,93 +1872,45 @@ static void dtmf_callback(pjmedia_stream *strm, void *user_data, } -pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, - const pjmedia_sdp_session *local_sdp, - const pjmedia_sdp_session *remote_sdp) +static pj_status_t audio_channel_update(pjsua_call_media *call_med, + pj_pool_t *tmp_pool, + 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; + pjsua_call *call = call_med->call; + pjmedia_stream_info the_si, *si = &the_si; pjmedia_port *media_port; + unsigned strm_idx = call_med->idx; pj_status_t status; - - if (!pjsua_var.med_endpt) { - /* We're being shutdown */ - return PJ_EBUSY; - } - - /* 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->pool_prov, - pjsua_var.med_endpt, - PJMEDIA_MAX_SDP_MEDIA, &sess_info, - local_sdp, remote_sdp); + + status = pjmedia_stream_info_from_sdp(si, tmp_pool, pjsua_var.med_endpt, + local_sdp, remote_sdp, strm_idx); if (status != PJ_SUCCESS) return status; - /* Update audio index from the negotiated SDP */ - call->audio_idx = find_audio_index(local_sdp, PJ_TRUE); - - /* 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)); - si = &sess_info.stream_info[0]; - } - /* Check if no media is active */ - if (sess_info.stream_cnt == 0 || si->dir == PJMEDIA_DIR_NONE) - { + if (si->dir == PJMEDIA_DIR_NONE) { /* Call media state */ - call->media_st = PJSUA_CALL_MEDIA_NONE; + call_med->state = PJSUA_CALL_MEDIA_NONE; /* Call media direction */ - call->media_dir = PJMEDIA_DIR_NONE; - - /* Don't stop transport because we need to transmit keep-alives, and - * also to prevent restarting ICE negotiation. See - * http://trac.pjsip.org/repos/ticket/1094 - */ -#if 0 - /* 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; - } -#endif + call_med->dir = PJMEDIA_DIR_NONE; } else { pjmedia_transport_info tp_info; /* Start/restart media transport */ - status = pjmedia_transport_media_start(call->med_tp, - call->inv->pool_prov, - local_sdp, remote_sdp, - call->audio_idx); + status = pjmedia_transport_media_start(call_med->tp, + tmp_pool, local_sdp, + remote_sdp, strm_idx); if (status != PJ_SUCCESS) return status; - call->med_tp_st = PJSUA_MED_TP_RUNNING; + 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); + pjmedia_transport_get_info(call_med->tp, &tp_info); if (tp_info.specific_info_cnt > 0) { unsigned i; for (i = 0; i < tp_info.specific_info_cnt; ++i) { @@ -1664,7 +1919,7 @@ pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, pjmedia_srtp_info *srtp_info = (pjmedia_srtp_info*) tp_info.spc_info[i].buffer; - call->rem_srtp_use = srtp_info->peer_use; + call_med->rem_srtp_use = srtp_info->peer_use; break; } } @@ -1693,16 +1948,16 @@ pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, si->jb_max = pjsua_var.media_cfg.jb_max; /* Set SSRC */ - si->ssrc = call->ssrc; + si->ssrc = call_med->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; + si->rtp_ts = call_med->rtp_tx_ts; + si->rtp_seq = call_med->rtp_tx_seq; + si->rtp_seq_ts_set = call_med->rtp_tx_seq_ts_set; #if defined(PJMEDIA_STREAM_ENABLE_KA) && PJMEDIA_STREAM_ENABLE_KA!=0 /* Enable/disable stream keep-alive and NAT hole punch. */ @@ -1710,9 +1965,15 @@ pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, #endif /* Create session based on session info. */ - status = pjmedia_session_create( pjsua_var.med_endpt, &sess_info, - &call->med_tp, - call, &call->session ); + status = pjmedia_stream_create(pjsua_var.med_endpt, NULL, si, + call_med->tp, NULL, + &call_med->strm.a.stream); + if (status != PJ_SUCCESS) { + return status; + } + + /* Start stream */ + status = pjmedia_stream_start(call_med->strm.a.stream); if (status != PJ_SUCCESS) { return status; } @@ -1721,23 +1982,24 @@ pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, * 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)); + pjmedia_stream_set_dtmf_callback(call_med->strm.a.stream, + &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); + pjmedia_stream_get_port(call_med->strm.a.stream, &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); + pjsua_var.ua_cfg.cb.on_stream_created(call->index, + call_med->strm.a.stream, + strm_idx, &media_port); } /* @@ -1758,66 +2020,132 @@ pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, call->inv->pool_prov, media_port, &port_name, - (unsigned*)&call->conf_slot); + (unsigned*) + &call_med->strm.a.conf_slot); if (status != PJ_SUCCESS) { return status; } } /* Call media direction */ - call->media_dir = si->dir; + call_med->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; + call_med->state = PJSUA_CALL_MEDIA_LOCAL_HOLD; + else if (call_med->dir == PJMEDIA_DIR_DECODING) + call_med->state = PJSUA_CALL_MEDIA_REMOTE_HOLD; else - call->media_st = PJSUA_CALL_MEDIA_ACTIVE; + call_med->state = PJSUA_CALL_MEDIA_ACTIVE; } /* Print info. */ { char info[80]; int info_len = 0; - unsigned i; + int len; + const char *dir; - for (i=0; i<sess_info.stream_cnt; ++i) { - int len; - const char *dir; - pjmedia_stream_info *strm_info = &sess_info.stream_info[i]; - - switch (strm_info->dir) { - 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; + switch (si->dir) { + 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)", strm_idx, + (int)si->fmt.encoding_name.slen, + si->fmt.encoding_name.ptr, + dir); + if (len > 0) + info_len += len; PJ_LOG(4,(THIS_FILE,"Media updates%s", info)); } return PJ_SUCCESS; } +pj_status_t pjsua_media_channel_update(pjsua_call_id call_id, + const pjmedia_sdp_session *local_sdp, + const pjmedia_sdp_session *remote_sdp) +{ + pjsua_call *call = &pjsua_var.calls[call_id]; + pj_pool_t *tmp_pool = call->inv->pool_prov; + unsigned mi; + pj_bool_t got_media = PJ_FALSE; + pj_status_t status = PJ_SUCCESS; + + if (pjsua_get_state() != PJSUA_STATE_RUNNING) + return PJ_EBUSY; + + /* Destroy existing media session, if any. */ + stop_media_session(call->index); + + /* Reset audio_idx first */ + call->audio_idx = -1; + + /* Process each media stream */ + for (mi=0; mi < call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; + + if (mi >= local_sdp->media_count || + mi >= remote_sdp->media_count) + { + /* This may happen when remote removed any SDP media lines in + * its re-offer. + */ + continue; +#if 0 + /* Something is wrong */ + PJ_LOG(1,(THIS_FILE, "Error updating media for call %d: " + "invalid media index %d in SDP", call_id, mi)); + return PJMEDIA_SDP_EINSDP; +#endif + } + + switch (call_med->type) { + case PJMEDIA_TYPE_AUDIO: + status = audio_channel_update(call_med, tmp_pool, + local_sdp, remote_sdp); + if (call->audio_idx==-1 && status==PJ_SUCCESS && + call_med->strm.a.stream) + { + call->audio_idx = mi; + } + break; +#if PJMEDIA_HAS_VIDEO + case PJMEDIA_TYPE_VIDEO: + status = video_channel_update(call_med, tmp_pool, + local_sdp, remote_sdp); + break; +#endif + default: + status = PJMEDIA_EINVALIMEDIATYPE; + break; + } + + if (status != PJ_SUCCESS) { + PJ_PERROR(1,(THIS_FILE, status, "Error updating media call%02d:%d", + call_id, mi)); + } else { + got_media = PJ_TRUE; + } + } + + return (got_media? PJ_SUCCESS : PJMEDIA_SDPNEG_ENOMEDIA); +} + /* * Get maxinum number of conference ports. */ @@ -1968,9 +2296,10 @@ PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, port0_info.listener_cnt==0 && port0_info.transmitter_cnt==0) { need_reopen = (peer_info.format.id != port0_info.format.id || - peer_info.format.bitrate != port0_info.format.bitrate || + peer_info.format.det.aud.avg_bps != + port0_info.format.det.aud.avg_bps || peer_info.clock_rate != port0_info.clock_rate || - peer_info.channel_count != port0_info.channel_count); + peer_info.channel_count!=port0_info.channel_count); } if (need_reopen) { @@ -1985,7 +2314,8 @@ PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, peer_info.samples_per_frame, peer_info.bits_per_sample); if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Error opening sound device", status); + pjsua_perror(THIS_FILE, "Error opening sound device", + status); return status; } @@ -1998,17 +2328,28 @@ PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, param.options = 0; status = open_snd_dev(¶m); if (status != PJ_SUCCESS) { - pjsua_perror(THIS_FILE, "Error opening sound device", status); + pjsua_perror(THIS_FILE, "Error opening sound device", + status); return status; } } else { /* Null-audio */ - status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); + 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); + pjsua_perror(THIS_FILE, "Error opening sound device", + status); return status; } } + } else if (pjsua_var.no_snd) { + if (!pjsua_var.snd_is_on) { + pjsua_var.snd_is_on = PJ_TRUE; + /* Notify app */ + if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { + (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); + } + } } } else { @@ -2025,8 +2366,13 @@ PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, pjsua_perror(THIS_FILE, "Error opening sound device", status); return status; } + } else if (pjsua_var.no_snd && !pjsua_var.snd_is_on) { + pjsua_var.snd_is_on = PJ_TRUE; + /* Notify app */ + if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { + (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); + } } - } return pjmedia_conf_connect_port(pjsua_var.mconf, source, sink, 0); @@ -2142,7 +2488,7 @@ PJ_DEF(pj_status_t) pjsua_player_create( const pj_str_t *filename, status = pjmedia_wav_player_port_create( pool, path, pjsua_var.mconf_cfg.samples_per_frame * - 1000 / pjsua_var.media_cfg.channel_count / + 1000 / pjsua_var.media_cfg.channel_count / pjsua_var.media_cfg.clock_rate, options, 0, &port); if (status != PJ_SUCCESS) { @@ -2683,6 +3029,11 @@ static pj_status_t open_snd_dev(pjmedia_snd_port_param *param) /* Close existing sound port */ close_snd_dev(); + /* Notify app */ + if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { + (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); + } + /* 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); @@ -2708,7 +3059,7 @@ static pj_status_t open_snd_dev(pjmedia_snd_port_param *param) */ if (!pjsua_var.is_mswitch && param->base.ext_fmt.id == PJMEDIA_FORMAT_PCM && - conf_port->info.clock_rate != param->base.clock_rate) + PJMEDIA_PIA_SRATE(&conf_port->info) != param->base.clock_rate) { pjmedia_port *resample_port; unsigned resample_opt = 0; @@ -2744,14 +3095,17 @@ static pj_status_t open_snd_dev(pjmedia_snd_port_param *param) * derived from the sound device setting, so update the setting. */ if (pjsua_var.is_mswitch) { - pj_memcpy(&conf_port->info.format, ¶m->base.ext_fmt, - sizeof(conf_port->info.format)); - conf_port->info.clock_rate = param->base.clock_rate; - conf_port->info.samples_per_frame = param->base.samples_per_frame; - conf_port->info.channel_count = param->base.channel_count; - conf_port->info.bits_per_sample = 16; + pj_memcpy(&conf_port->info.fmt, ¶m->base.ext_fmt, + sizeof(conf_port->info.fmt)); + conf_port->info.fmt.det.aud.clock_rate = param->base.clock_rate; + conf_port->info.fmt.det.aud.frame_time_usec = param->base.samples_per_frame* + 1000000 / + param->base.clock_rate; + conf_port->info.fmt.det.aud.channel_count = param->base.channel_count; + conf_port->info.fmt.det.aud.bits_per_sample = 16; } + /* Connect sound port to the bridge */ status = pjmedia_snd_port_connect(pjsua_var.snd_port, conf_port ); @@ -2816,6 +3170,11 @@ static pj_status_t open_snd_dev(pjmedia_snd_port_param *param) /* Close existing sound device */ static void close_snd_dev(void) { + /* Notify app */ + if (pjsua_var.snd_is_on && pjsua_var.ua_cfg.cb.on_snd_dev_operation) { + (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(0); + } + /* Close sound device */ if (pjsua_var.snd_port) { pjmedia_aud_dev_info cap_info, play_info; @@ -2849,6 +3208,7 @@ static void close_snd_dev(void) if (pjsua_var.snd_pool) pj_pool_release(pjsua_var.snd_pool); pjsua_var.snd_pool = NULL; + pjsua_var.snd_is_on = PJ_FALSE; } @@ -2912,6 +3272,7 @@ PJ_DEF(pj_status_t) pjsua_set_snd_dev( int capture_dev, } pjsua_var.no_snd = PJ_FALSE; + pjsua_var.snd_is_on = PJ_TRUE; return PJ_SUCCESS; } @@ -2947,6 +3308,11 @@ PJ_DEF(pj_status_t) pjsua_set_null_snd_dev(void) /* Close existing sound device */ close_snd_dev(); + /* Notify app */ + if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { + (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); + } + /* 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); @@ -2976,6 +3342,7 @@ PJ_DEF(pj_status_t) pjsua_set_null_snd_dev(void) pjsua_var.play_dev = NULL_SND_DEV_ID; pjsua_var.no_snd = PJ_FALSE; + pjsua_var.snd_is_on = PJ_TRUE; return PJ_SUCCESS; } @@ -3121,6 +3488,8 @@ PJ_DEF(pj_status_t) pjsua_enum_codecs( pjsua_codec_info id[], if (count > *p_count) count = *p_count; for (i=0; i<count; ++i) { + pj_bzero(&id[i], sizeof(pjsua_codec_info)); + pjmedia_codec_info_to_id(&info[i], id[i].buf_, sizeof(id[i].buf_)); id[i].codec_id = pj_str(id[i].buf_); id[i].priority = (pj_uint8_t) prio[i]; @@ -3174,7 +3543,7 @@ PJ_DEF(pj_status_t) pjsua_codec_get_param( const pj_str_t *codec_id, return status; if (count != 1) - return PJ_ENOTFOUND; + return (count > 1? PJ_ETOOMANY : PJ_ENOTFOUND); status = pjmedia_codec_mgr_get_default_param( codec_mgr, info, param); return status; @@ -3211,3 +3580,5 @@ PJ_DEF(pj_status_t) pjsua_codec_set_param( const pj_str_t *codec_id, status = pjmedia_codec_mgr_set_default_param(codec_mgr, info[0], param); return status; } + + |