diff options
Diffstat (limited to 'pjsip/src/pjsua-lib/pjsua_media.c')
| -rw-r--r-- | pjsip/src/pjsua-lib/pjsua_media.c | 55 |
1 files changed, 48 insertions, 7 deletions
diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c index e7743659..60aebbeb 100644 --- a/pjsip/src/pjsua-lib/pjsua_media.c +++ b/pjsip/src/pjsua-lib/pjsua_media.c @@ -396,7 +396,7 @@ pj_status_t pjsua_media_subsys_start(void) /* * Destroy pjsua media subsystem. */ -pj_status_t pjsua_media_subsys_destroy(void) +pj_status_t pjsua_media_subsys_destroy(unsigned flags) { unsigned i; @@ -441,6 +441,10 @@ pj_status_t pjsua_media_subsys_destroy(void) pjsua_media_channel_deinit(i); } if (call_med->tp && call_med->tp_auto_del) { + /* TODO: check if we're not allowed to send to network in the + * "flags", and if so do not do TURN allocation... + */ + PJ_UNUSED_ARG(flags); pjmedia_transport_close(call_med->tp); } call_med->tp = NULL; @@ -1294,11 +1298,18 @@ static pj_status_t call_media_init_cb(pjsua_call_media *call_med, if (call_med->tp_st == PJSUA_MED_TP_CREATING) set_media_tp_state(call_med, PJSUA_MED_TP_IDLE); + if (!call_med->tp_orig && + pjsua_var.ua_cfg.cb.on_create_media_transport) + { + call_med->use_custom_med_tp = PJ_TRUE; + } else + call_med->use_custom_med_tp = PJ_FALSE; + #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->tp_orig || call_med->tp == call_med->tp_orig) { + if (!call_med->tp_orig) { pjmedia_srtp_setting srtp_opt; pjmedia_transport *srtp = NULL; @@ -1314,7 +1325,7 @@ static pj_status_t call_media_init_cb(pjsua_call_media *call_med, /* Always create SRTP adapter */ pjmedia_srtp_setting_default(&srtp_opt); srtp_opt.close_member_tp = PJ_TRUE; - /* If media session has been ever established, let's use remote's + /* If media session has been ever established, let's use remote's * preference in SRTP usage policy, especially when it is stricter. */ if (call_med->rem_srtp_use > acc->cfg.use_srtp) @@ -1519,9 +1530,25 @@ static pj_status_t media_channel_init_cb(pjsua_call_id call_id, call->async_call.dlg->pool); } - status = pjmedia_transport_media_create( - call_med->tp, tmp_pool, - 0, call->async_call.rem_sdp, mi); + if (call_med->use_custom_med_tp) { + unsigned custom_med_tp_flags = 0; + + /* Use custom media transport returned by the application */ + call_med->tp = + (*pjsua_var.ua_cfg.cb.on_create_media_transport) + (call_id, mi, call_med->tp, + custom_med_tp_flags); + if (!call_med->tp) { + status = + PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE); + } + } + + if (call_med->tp) { + status = pjmedia_transport_media_create( + call_med->tp, tmp_pool, + 0, call->async_call.rem_sdp, mi); + } if (status != PJ_SUCCESS) { call->med_ch_info.status = status; call->med_ch_info.med_idx = mi; @@ -2104,6 +2131,7 @@ static void stop_media_session(pjsua_call_id call_id) PJ_LOG(4,(THIS_FILE, "Media session call%02d:%d is destroyed", call_id, mi)); + call_med->prev_state = call_med->state; call_med->state = PJSUA_CALL_MEDIA_NONE; } @@ -2133,12 +2161,19 @@ pj_status_t pjsua_media_channel_deinit(pjsua_call_id call_id) PJ_LOG(4,(THIS_FILE, "Call %d: deinitializing media..", call_id)); pj_log_push_indent(); + for (mi=0; mi<call->med_cnt; ++mi) { + pjsua_call_media *call_med = &call->media[mi]; + + if (call_med->type == PJMEDIA_TYPE_AUDIO && call_med->strm.a.stream) + pjmedia_stream_send_rtcp_bye(call_med->strm.a.stream); + } + stop_media_session(call_id); for (mi=0; mi<call->med_cnt; ++mi) { pjsua_call_media *call_med = &call->media[mi]; - if (call_med->tp_st > PJSUA_MED_TP_IDLE) { + if (call_med->tp_st > PJSUA_MED_TP_IDLE) { pjmedia_transport_media_stop(call_med->tp); set_media_tp_state(call_med, PJSUA_MED_TP_IDLE); } @@ -2153,6 +2188,7 @@ pj_status_t pjsua_media_channel_deinit(pjsua_call_id call_id) pjmedia_transport_close(call_med->tp); call_med->tp = call_med->tp_orig = NULL; } + call_med->tp_orig = NULL; } check_snd_dev_idle(); @@ -2206,6 +2242,8 @@ static pj_status_t audio_channel_update(pjsua_call_media *call_med, if (status != PJ_SUCCESS) goto on_return; + si->rtcp_sdes_bye_disabled = PJ_TRUE; + /* Check if no media is active */ if (si->dir == PJMEDIA_DIR_NONE) { /* Call media state */ @@ -2296,6 +2334,9 @@ static pj_status_t audio_channel_update(pjsua_call_media *call_med, goto on_return; } + if (call_med->prev_state == PJSUA_CALL_MEDIA_NONE) + pjmedia_stream_send_rtcp_sdes(call_med->strm.a.stream); + /* If DTMF callback is installed by application, install our * callback to the session. */ |