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SIP/media Features About: PJLIB, PJLIB-UTIL, PJSIP, and PJMEDIA are created by: Benny Prijono <bennylp  pjsip.org>
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Home --> Documentations --> PJSIP Reference pjsua_media_config Struct Reference |
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Data Fields | |
| unsigned | clock_rate |
| unsigned | snd_clock_rate |
| unsigned | channel_count |
| unsigned | audio_frame_ptime |
| unsigned | max_media_ports |
| pj_bool_t | has_ioqueue |
| unsigned | thread_cnt |
| unsigned | quality |
| unsigned | ptime |
| pj_bool_t | no_vad |
| unsigned | ilbc_mode |
| unsigned | tx_drop_pct |
| unsigned | rx_drop_pct |
| unsigned | ec_options |
| unsigned | ec_tail_len |
| unsigned | snd_rec_latency |
| unsigned | snd_play_latency |
| int | jb_init |
| int | jb_min_pre |
| int | jb_max_pre |
| int | jb_max |
| pj_bool_t | enable_ice |
| int | ice_max_host_cands |
| pj_ice_sess_options | ice_opt |
| pj_bool_t | ice_no_rtcp |
| pj_bool_t | enable_turn |
| pj_str_t | turn_server |
| pj_turn_tp_type | turn_conn_type |
| pj_stun_auth_cred | turn_auth_cred |
| int | snd_auto_close_time |
Field Documentation
| unsigned pjsua_media_config::clock_rate |
Clock rate to be applied to the conference bridge. If value is zero, default clock rate will be used (PJSUA_DEFAULT_CLOCK_RATE, which by default is 16KHz).
| unsigned pjsua_media_config::snd_clock_rate |
Clock rate to be applied when opening the sound device. If value is zero, conference bridge clock rate will be used.
| unsigned pjsua_media_config::channel_count |
Channel count be applied when opening the sound device and conference bridge.
Specify audio frame ptime. The value here will affect the samples per frame of both the sound device and the conference bridge. Specifying lower ptime will normally reduce the latency.
Default value: PJSUA_DEFAULT_AUDIO_FRAME_PTIME
| unsigned pjsua_media_config::max_media_ports |
Specify maximum number of media ports to be created in the conference bridge. Since all media terminate in the bridge (calls, file player, file recorder, etc), the value must be large enough to support all of them. However, the larger the value, the more computations are performed.
Default value: PJSUA_MAX_CONF_PORTS
Specify whether the media manager should manage its own ioqueue for the RTP/RTCP sockets. If yes, ioqueue will be created and at least one worker thread will be created too. If no, the RTP/RTCP sockets will share the same ioqueue as SIP sockets, and no worker thread is needed.
Normally application would say yes here, unless it wants to run everything from a single thread.
| unsigned pjsua_media_config::thread_cnt |
Specify the number of worker threads to handle incoming RTP packets. A value of one is recommended for most applications.
| unsigned pjsua_media_config::quality |
Media quality, 0-10, according to this table: 5-10: resampling use large filter, 3-4: resampling use small filter, 1-2: resampling use linear. The media quality also sets speex codec quality/complexity to the number.
Default: 5 (PJSUA_DEFAULT_CODEC_QUALITY).
| unsigned pjsua_media_config::ptime |
Specify default codec ptime.
Default: 0 (codec specific)
Disable VAD?
Default: 0 (no (meaning VAD is enabled))
| unsigned pjsua_media_config::ilbc_mode |
iLBC mode (20 or 30).
Default: 30 (PJSUA_DEFAULT_ILBC_MODE)
| unsigned pjsua_media_config::tx_drop_pct |
Percentage of RTP packet to drop in TX direction (to simulate packet lost).
Default: 0
| unsigned pjsua_media_config::rx_drop_pct |
Percentage of RTP packet to drop in RX direction (to simulate packet lost).
Default: 0
| unsigned pjsua_media_config::ec_options |
Echo canceller options (see pjmedia_echo_create())
Default: 0.
| unsigned pjsua_media_config::ec_tail_len |
Echo canceller tail length, in miliseconds.
Default: PJSUA_DEFAULT_EC_TAIL_LEN
| unsigned pjsua_media_config::snd_rec_latency |
Audio capture buffer length, in milliseconds.
Default: PJMEDIA_SND_DEFAULT_REC_LATENCY
| unsigned pjsua_media_config::snd_play_latency |
Audio playback buffer length, in milliseconds.
Default: PJMEDIA_SND_DEFAULT_PLAY_LATENCY
Jitter buffer initial prefetch delay in msec. The value must be between jb_min_pre and jb_max_pre below.
Default: -1 (to use default stream settings, currently 150 msec)
Jitter buffer minimum prefetch delay in msec.
Default: -1 (to use default stream settings, currently 60 msec)
Jitter buffer maximum prefetch delay in msec.
Default: -1 (to use default stream settings, currently 240 msec)
Set maximum delay that can be accomodated by the jitter buffer msec.
Default: -1 (to use default stream settings, currently 360 msec)
Enable ICE
Set the maximum number of host candidates.
Default: -1 (maximum not set)
ICE session options.
Disable RTCP component.
Default: no
Enable TURN relay candidate in ICE.
Specify TURN domain name or host name, in in "DOMAIN:PORT" or "HOST:PORT" format.
Specify the connection type to be used to the TURN server. Valid values are PJ_TURN_TP_UDP or PJ_TURN_TP_TCP.
Default: PJ_TURN_TP_UDP
Specify the credential to authenticate with the TURN server.
Specify idle time of sound device before it is automatically closed, in seconds. Use value -1 to disable the auto-close feature of sound device
Default : 1
The documentation for this struct was generated from the following file:
Copyright (C) 2006-2008 Teluu Inc.