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Asterisk, since its early days, has offered a conferencing application called MeetMe (app_meetme.so). MeetMe provides DAHDI-mixed software-based bridges for multi-party audio conferencing. MeetMe is used by nearly all Asterisk implementations - small office, call center, large office, feature-server, third-party application, etc. It has been extremely successful as an audio bridge.
Over time, several significant limitations of MeetMe have been encountered by its users. Among these are two of distinction: MeetMe requires DAHDI for mixing, and is thus limited to 8kHz (PSTN) audio sampling rates; and MeetMe is delivered in a fairly static form, it does not provide extensive configuration options.
To address these limitations, a new conferencing application, based upon the ConfBridge application introduced in Asterisk 1.6.0, is now available with Asterisk 10. This new ConfBridge application replaces the older ConfBridge application. It is not intended to be a direct replacement for MeetMe, it will not provide feature parity with the MeetMe application. Instead, the new ConfBridge application delivers a completely redesigned set of functionality that most users will find more than sufficient, and in many ways better, for their conferencing needs.
ConfBridge provides four internal concepts:
- Conference Number
- Bridge Profile
- User Profile
- Conference Menu
A Conference Number is a numerical representation for an instance of the bridge. Callers joined to the same conference number will be in the same conference bridge; they're connected. Callers joined to different conference numbers are not in the same conference bridge; they're separated. Conference Numbers are assigned in the dialplan. Unlike MeetMe, they're not pre-reserved.
A Bridge Profile is a named set of options that control the behavior of a particular conference bridge. Each bridge must have its own profile. A single bridge cannot have more than one Bridge Profile.
A User Profile is a named set of options that control the user's experience as a member of a particular bridge. Each user participating in a bridge can have their own individual User Profile.
A Conference Menu is a named set of options that are provided to a user when they present DTMF keys while connected to the bridge. Each user participating in a bridge can have their own individual Conference Menu.
ConfBridge Profiles and Menus are configured in the confbridge.conf configuration file - normally located at /etc/asterisk/confbridge.conf. The file contains three reserved sections:
The [general] section is currently unused, but is reserved for future use.
The [default_bridge] section contains all options invoked when ConfBridge is instantiated from the dialplan without a bridge profile argument.
The [default_user] section contains all options invoked when ConfBridge is instantiated from the dialplan without a user profile argument.
Each section contains a type definition. The type definition determines the function of the section. The three types are:
bridge is used to denote Bridge Profile section definitions.
user is used to denote User Profile section definitions.
menu is used to denote Conference Menu section definitions.
All other sections, defined by a section identifier encapsulated in square brackets, are user-definable.
This is an example, using invalid options and functions, of a confbridge.conf configuration file, displaying the organizational layout. The various options and functions are described later in this page.
[general] ; comments are preceded by a comma ; ; the general section is blank ; [default_bridge] type=bridge ; Bridge Profile options go here myoption=value myoption2=othervalue ; [default_user] type=user ; User Profile options go here myoption=value myoption2=othervalue ; [sample_menu] type=menu ; Conferece Menu options go here DTMF=function otherDTMF=otherFunction ;
Bridge Profile Configuration Options
A Bridge Profile provides the following configuration options:
Set this to bridge to configure a bridge profile
integer; e.g. 50
Limits the number of participants for a single conference to a specific number. By default, conferences have no participant limit. After the limit is reached, the conference will be locked until someone leaves. Admin-level users are exempt from this limit and will still be able to join otherwise-locked, because of limit, conferences.
Records the conference call starting when the first user enters the room, and ending when the last user exits the room. The default recorded filename is 'confbridge-<name of conference bridge>-<start time>.wav and the default format is 8kHz signed linear. By default, this option is disabled. This file will be located in the configured monitoring directory as set in asterisk.conf
path, e.g. /tmp/myfiles
When record_conference is set to yes, the specific name of the recorded file can be set using this option. Note that since multiple conferences may use the same Bridge profile, this can cause issues, depending on the configuration. It is recommended to only use this option dynamically with the CONFBRIDGE() dialplan function. This allows the recorded name to be specified and a unique name to be chosen. By default, the recorded file is stored in Asterisk's spool/monitory directory, with a unique filename starting with the 'confbridge' prefix.
auto, 8000, 12000, 16000, 24000, 32000, 44100, 48000, 96000, 192000
Sets the internal native sample rate at which to mix the conference. The "auto" option allows Asterisk to adjust the sample rate to the best quality / performance based on the participant makeup. Numbered values lock the rate to the specified numerical rate. If a defined number does not match an internal sampling rate supported by Asterisk, the nearest sampling rate will be used instead.
10, 20, 40, 80
Sets, in milliseconds, the internal mixing interval. By default, the mixing interval of a bridge is 20ms. This setting reflects how "tight" or "loose" the mixing will be for the conference. Lower intervals provide a "tighter" sound with less delay in the bridge and consume more system resources. Higher intervals provide a "looser" sound with more delay in the bridge and consume less resources
none, follow_talker, last_marked, first_marked
Configured video (as opposed to audio) distribution method for conference participants. Participants must use the same video codec. Confbridge does not provide MCU functionality. It does not transcode, scale, transrate, or otherwise manipulate the video. Options are "none," where no video source is set by default and a video source may be later set via AMI or DTMF actions; "follow_talker," where video distrubtion follows whomever is talking and providing video; "last_marked," where the last marked user with video capabilities to join the conference will be the single video source distributed to all other participants - when the current video source leaves, the marked user previous to the last-joined will be used as the video source; and "first-marked," where the first marked user with video capabilities to join the conference will be the single video source distributed to all other participants - when the current video source leaves, the marked user that joined next will be used as the video source. Use of video in conjunction with the jitterbuffer results in the audio being slightly out of sync with the video - because the jitterbuffer only operates on the audio stream, not the video stream. Jitterbuffer should be disabled when video is used.
The sound played to the bridge when a user joins, typically some kind of beep sound
The sound played to the bridge when a user leaves, also typically some kind of beep sound
The sound played as a user intro, e.g. "xxxx has joined the conference."
The sound played as a user parts the conference, e.g. "xxxx has left the conference."
The sound played to a user who has been kicked from the conference.
The sound played to a user when the mute option is toggled on.
The sound played to a user when the mute option is toggled off.
The sound played when a user is the only person in the conference.
The sound played to a user when there is only one other person in the conference.
The sound played when announcing how many users there are in a conference.
Used in conjunction with the sound_there_are option, used like "sound_there_are" <number of participants> "sound_other_in_party"
The sound played when someone is placed into a conference, after waiting for a marked user.
The sound played when a user is placed into a conference that cannot start until a marked user enters.
The sound played when the last marked user leaves the conference.
The sound played when prompting for a conference PIN
The sound played when an invalid PIN is entered too many (3) times
The sound played to a user trying to join a locked conference.
The sound played to an Admin-level user after toggling the conference to locked mode.
The sound played to an Admin-level user after toggling the conference to unlocked mode.
The sound played when an invalid menu option is entered.
In this example, a Bridge Profile called "fancybridge" will be created. It will be configured to allow up to 20 callers, and will be set to mix at 10ms (tight mixing) at an automatic sampling rate. Additionally, it will be recorded.
[fancybridge] type=bridge max_members=20 mixing_interval=10 internal_sample_rate=auto record_conference=yes
User Profile Configuration Options
A User Profile provides the following configuration options:
Set this to user to configure a user profile
Sets if the user is an Admin or not. By default, no.
Sets if the user is Marked or not. By default, no.
sets if the user should start out muted. By default, no.
Sets whether music on hold should be played when only one person is in the conference or when the user is waiting on a marked user to enter the conference. By default, off.
music on hold class
Sets the music on hold class to use for music on hold.
When set to "yes," enter/leave prompts and user introductions are not played. By default, no.
Sets if the number of users in the conference should be announced to the caller. By default, no.
yes/no; or an integer
Sets if the number of users should be announced to all other users in the conference when someone joins. When set to a number, the announcement will only occur once the user count is above the specified number
Sets if the only user announcement should be played when someone enters an empty conference. By default, yes.
Sets if the user must wait for another marked user to enter before joining the conference. By default, no.
If enabled, every user with this option in their profile will be removed from the conference when the last marked user exists the conference.
Drops what Asterisk detects as silence from entering into the bridge. Enabling this option will drastically improve performance and help remove the buildup of background noise from the conference. This option is highly recommended for large conferences, due to its performance improvements.
integer in milliseconds
The time, in milliseconds, by default 160, of sound above what the DSP has established as base-line silence for a user, before that user is considered to be talking. This value affects several options:
integer in milliseconds
The time, in milliseconds, by default 2500, of sound falling within what the DSP has established as the baseline silence, before a user is considered to be silent. The best way to approach this option is to set it slightly above the maximum amount of milliseconds of silence a user may generate during natural speech. This value affects several operations:
Sets whether or not notifications of when a user begins and ends talking should be sent out as events over AMI. By default, no.
Whether or not a noise reduction filter should be applied to the audio before mixing. By default, off. This requires codec_speex to be built and installed. Do not confuse this option with drop_silence. denoise is useful if there is a lot of background noise for a user, as it attempts to remove the noise while still preserving the speech. This option does not remove silence from being mixed into the conference and does come at the cost of a slight performance hit.
Whether or not to place a jitter buffer on the caller's audio stream before any audio mixing is performed. This option is highly recommended, but will add a slight delay to the audio and will incur a slight performance penalty. This option makes use of the JITTERBUFFER dialplan function's default adaptive jitter buffer. For a more fine-tuned jitter buffer, disable this option and use the JITTERBUFFER dialplan function on the calling channel, before it enters the ConfBridge application.
Sets if the user must enter a PIN before joining the conference. The user will be prompted for the PIN.
When enabled, this option prompts the user for their name when entering the conference. After the name is recorded, it will be played as the user enters and exists the conference. By default, no.
Whether or not DTMF received from users should pass through the conference to other users. By default, no.
In this example, we will create a user profile called "fancyuser" that includes music on hold, user count announcements, join/leave announcements, silence detection, noise reduction and requires a PIN of 456.
[fancyuser] type=user music_on_hold_when_empty=yes music_on_hold_class=default announce_user_count_all=yes announce_join_leave=yes dsp_drop_silence=yes denoise=yes pin=456
Conference Menu Configuration Options
A Conference Menu provides the following configuration options:
Set this to menu to configure a conference menu
(<name of audio file1>&<name of audio file2>&...)
Plays back an audio file, or a string of audio files chained together using the & character, to the user and then immediately returns them to the conference.
(<name of audio file 1>&<name of audio file 2>&...)
Plays back an audio file, or a series of audio files chained together using the & character, to the user while continuing the collect the DTMF sequence. This is useful when using a menu prompt that describes all of the menu options. Note that any DTMF during this action will terminate the prompt's playback.
Toggles mute on and off. When a user is muted, they will not be able to speak to other conference users, but they can still listen to other users. While muted, DTMF keys from the caller will continue to be collected.
This action does nothing. Its only real purpose exists for being able to reserve a sequence in the configuration as a menu exit sequence.
Decreases the caller's listening volume. Everything they hear will sound quieter.
Increases the caller's listening volume. Everything they hear will sound louder.
Resets the caller's listening volume to the default level.
Decreases the caller's talking volume. Everything they say will sound quieter to other callers.
Increases the caller's talking volume. Everything they say will sound louder to other callers.
Resets the caller's talking volume to the default level.
Allows one to escape from the conference and execute commands in the dialplan. Once the dialplan exits, the user will be put back into the conference.
Allows a user to exit the conference and continue execution in the dialplan.
Allows an Admin to remove the last participant from the conference. This action only works for users whose User Profiles set them as conference Admins.
Allows an Admin to toggle locking and unlocking the conference. When the conference is locked, only other Admin users can join. When the conference is unlocked, any user may join up to the limit defined by the max_members Bridge Profile option. This action only works for users whose User Profiles set them as conference Admins.
Allows an Admin to mute/unmute all non-admin participants in the conference.
New in Asterisk 11
Allows a user to set themselves as the single video distribution source for all other participants. This overrides the video_mode setting.
Allows a user to release themselves as the single video source. Upon release of the video source, and/or if video_mode is set to "none," this action will result in the conference returning to whatever video mode the Bridge Profile is using. This action will have no effect if the user is not currently the video source. The user is also not guaranteed that the use of this action will prevent them from becoming the video source later.
In this example, we'll create a menu called "fancymenu." This menu will utilize many of the options listed above. We will construct a features menu that plays when the user enters the * character. Since we will do this using the playback_and_continue option, we will define other menu items as being a "subset" of the * command, e.g. *4, so that once the user presses *, they can listen to the menu options and then press the specific "after-star" option, e.g. 4, to affect the option. Additionally, we will duplicate those same sub-features as non-* features, so that the user does not need to have entered the * menu structure in order to affect the options, they can just press the key, e.g. "4" at any time, regardless of whether or not they're in the *-tree.
[fancymenu] type=menu *=playback_and_continue(conf-togglemute&press&digits/1&silence/1&conf-leave&press&digits/2&silence/1&add-a-caller&press&digits/3&silence/1&conf-decrease-talking&press&digits/4&silence/1&reset-talking&press&digits/5&silence/1&increase-talking&press&digits/6&silence/1&conf-decrease-listening&press&digits/7&silence/1&conf-reset-listening&press&digits/8&silence/1&conf-increase-listening&press&digits/9&silence/1&conf-exit-menu&press&digits/0) *1=toggle_mute 1=toggle_mute *2=leave_conference 2=leave_conference *3=dialplan_exec(addcallers,1,1) 3=dialplan_exec(addcallers,1,1) *4=decrease_listening_volume 4=decrease_listening_volume *5=reset_listening_volume 5=reset_listening_volume *6=increase_listening_volume 6=increase_listening_volume *7=decrease_talking_volume 7=decrease_talking_volume *8=reset_talking_volume 8=reset_talking_volume *9=increase_talking_volume 9=increase_talking_volume *0=no_op 0=no_op
Of particular note in this example, we're calling the dialplan_exec option. Here, we're specifying "addcaller,1,1." This means that when someone dials 3, Asterisk will escape them out of the bridge momentarily to go execute priority 1 of extension 1 in the addcaller context of the dialplan (extensions.conf). Our dialplan, including the addcaller context, in this case, might look like:
[addcaller] exten => 1,1,Originate(SIP/otherpeer,exten,conferences,100,1) [conferences] exten => 100,1,ConfBridge(1234)
Thus, when someone dials "3" while in the bridge, they'll Originate a call from the dialplan that puts SIP/otherpeer into the conference. Once the dial has completed, the person that dialed "3" will find themselves back in the bridge, with the other participants.
ConfBridge Dialplan Syntax
The syntax for the new ConfBridge application is as follows:
The ConfBridge application takes the following arguments
- confno - The conference number
- bridge_profile - The Bridge Brofile name from confbridge.conf. When left blank, a dynamically built Bridge Profile created by the CONFBRIDGE dialplan function is searched for on the channel and, if available, used. If no dynamic profile is found, the "default_bridge" profile found in confbridge.conf is used.
It is important to note that while User Profiles are unique for each participant, Bridge Profiles are unique to the bridge, not the user. So you can only create one Bridge Profile per conference
- user_profile - The User Profile name from confbridge.conf. When left blank, a dynamically built User Profile created by the CONFBRIDGE dialplan function is searched for on the channel and, if available, used. If no dynamic profile is present, the "default_user" profile found in confbridge.conf is used.
- menu - The Conference Menu name from confbridge.conf. No menu is applied by default if this option is left blank.
The ConfBridge application syntax and usage can be found at Asterisk 13 Application_ConfBridge
ConfBridge Application Examples
In this example, callers will be joined to conference number 1234, using the default Bridge Profile, the default User Profile, and no Conference Menu.
exten => 1,1,Answer() exten => 1,n,ConfBridge(1234)
In this example, callers will be joined to conference number 1234, with the default Bridge Profile, a User Profile called "1234_participants" and a Conference Menu called "1234_menu."
exten => 1,1,Answer() exten => 1,n,ConfBridge(1234,,1234_participants,1234_menu)
ConfBridge Dialplan Functions
The CONFBRIDGE dialplan function is used to set customized Bridge and/or User Profiles on a channel for the ConfBridge application. It uses the same options defined in confbridge.conf and allows the creation of dynamic, dialplan-driven conferences.
- type - Refers to which type of profile the option belongs to. Type can be either "bridge" or "user."
- option - Refers to the confbridge.conf option that is to be set dynamically on the channel. This can also refer to an existing Bridge or User Profile by using the keyword "template." In this case, an existing Bridge or User Profile can be appended or modified on-the-fly.
In this example, the custom set User Profile on this channel enables announce_join_leave (so users will be announced as they come and go), sets users to join muted (so that they're not able to speak), and pushes them into bridge "1."
exten => 1,1,Answer() exten => 1,n,Set(CONFBRIDGE(user,announce_join_leave)=yes) exten => 1,n,Set(CONFBRIDGE(user,startmuted)=yes) exten => 1,n,ConfBridge(1)
In this example, we will include an existing User Profile, the default_user User Profile as defined in confbridge.comf, and we will set additional parameters (admin and marked) that aren't already defined in the default_user User Profile.
exten => 1,1,Answer() exten => 1,n,Set(CONFBRIDGE(user,template)=default_user) exten => 1,n,Set(CONFBRIDGE(user,admin)=yes) exten => 1,n,Set(CONFBRIDGE(user,marked)=yes) exten => 1,n,ConfBridge(1)
The CONFBRIDGE_INFO dialplan function is used to retrieve information about a conference, such as locked/unlocked status and the number of parties including admins and marked users.
- type - Refers to which information type to be retrieved. Type can be either "parties," "admins," "marked," or "locked."
- conf - Refers to the name of the conference being referenced.
The CONFBRIDGE_INFO function returns a non-negative integer for valid conference identifiers, 0 or 1 for locked, and "" for invalid conference identifiers.
ConfBridge CLI Options
ConfBridge offers several options that may be invoked from the Asterisk CLI.
confbridge kick <conference> <channel>
Removes the specified channel from the conference, e.g.:
*CLI> confbridge kick 1111 SIP/mypeer-00000000 Kicking SIP/mypeer-00000000 from confbridge 1111
Shows a summary listing of all bridges, e.g.:
*CLI> confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== 1111 1 0 unlocked
confbridge list <conference>
Shows a detailed listing of participants in a specified conference, e.g.:
*CLI> confbridge list 1111 Channel User Profile Bridge Profile Menu ============================= ================ ================ ================ SIP/mypeer-00000001 default_user 1111 sample_user_menu
confbridge lock <conference>
Locks a specified conference so that only Admin users can join, e.g.:
*CLI> confbridge lock 1111 Conference 1111 is locked.
confbridge unlock <conference>
Unlocks a specified conference so that only Admin users can join, e.g.:
*CLI> confbridge unlock 1111 Conference 1111 is unlocked.
confbridge mute <conference> <channel>
Mutes a specified user in a specified conference, e.g.:
*CLI> confbridge mute 1111 SIP/mypeer-00000001 Muting SIP/mypeer-00000001 from confbridge 1111
confbridge unmute <conference> <channel>
Unmutes a specified user in a specified conference, e.g.:
*CLI> confbridge unmute 1111 SIP/mypeer-00000001 Unmuting SIP/mypeer-00000001 from confbridge 1111
confbridge record start <conference> <file>
*CLI> confbridge record start 1111 Recording started *CLI> == Begin MixMonitor Recording ConfBridgeRecorder/conf-1111-uid-618880445
confbridge record stop <confererence>
Stops recording the specified conference, e.g.:
*CLI> confbridge record stop 1111 Recording stopped. *CLI> == MixMonitor close filestream (mixed) == End MixMonitor Recording ConfBridgeRecorder/conf-1111-uid-618880445
confbridge show menus
Shows a listing of Conference Menus as defined in confbridge.conf, e.g.:
*CLI> confbridge show menus --------- Menus ----------- sample_admin_menu sample_user_menu
confbridge show menu <menu name>
Shows a detailed listing of a named Conference Menu, e.g.:
*CLI> confbridge show menu sample_admin_menu Name: sample_admin_menu *9=increase_talking_volume *8=no_op *7=decrease_talking_volume *6=increase_listening_volume *4=decrease_listening_volume *3=admin_kick_last *2=admin_toggle_conference_lock *1=toggle_mute *=playback_and_continue(conf-adminmenu)
confbridge show profile bridges
Shows a listing of Bridge Profiles as defined in confbridge.conf, e.g.:
*CLI> confbridge show profile bridges --------- Bridge Profiles ----------- 1111 default_bridge
confbridge show profile bridge <bridge>
Shows a detailed listing of a named Bridge Profile, e.g.:
*CLI> confbridge show profile bridge 1111 -------------------------------------------- Name: 1111 Internal Sample Rate: 16000 Mixing Interval: 10 Record Conference: no Record File: Auto Generated Max Members: No Limit sound_only_person: conf-onlyperson sound_has_joined: conf-hasjoin sound_has_left: conf-hasleft sound_kicked: conf-kicked sound_muted: conf-muted sound_unmuted: conf-unmuted sound_there_are: conf-thereare sound_other_in_party: conf-otherinparty sound_place_into_conference: conf-placeintoconf sound_wait_for_leader: conf-waitforleader sound_get_pin: conf-getpin sound_invalid_pin: conf-invalidpin sound_locked: conf-locked sound_unlocked_now: conf-unlockednow sound_lockednow: conf-lockednow sound_error_menu: conf-errormenu
confbridge show profile users
Shows a listing of User Profiles as defined in confbridge.conf, e.g.:
*CLI> confbridge show profile users --------- User Profiles ----------- awesomeusers default_user
confbirdge show profile user <user>
Shows a detailed listing of a named Bridge Profile, e.g.:
*CLI> confbridge show profile user default_user -------------------------------------------- Name: default_user Admin: false Marked User: false Start Muted: false MOH When Empty: enabled MOH Class: default Quiet: disabled Wait Marked: disabled END Marked: disabled Drop_silence: enabled Silence Threshold: 2500ms Talking Threshold: 160ms Denoise: disabled Talk Detect Events: disabled DTMF Pass Through: disabled PIN: None Announce User Count: enabled Announce join/leave: enabled Announce User Count all: enabled
ConfBridge Asterisk Manager Interface (AMI) Actions
Lists all users in a particular ConfBridge conference. ConfbridgeList will follow as separate events, followed by a final event called ConfbridgeListComplete
Action: ConfbridgeList Conference: 1111 Response: Success EventList: start Message: Confbridge user list will follow Event: ConfbridgeList Conference: 1111 CallerIDNum: malcolm CallerIDName: malcolm Channel: SIP/malcolm-00000000 Admin: No MarkedUser: No Event: ConfbridgeListComplete EventList: Complete ListItems: 1
Lists data about all active conferences. ConfbridgeListRooms will follow as separate events, followed by a final event called ConfbridgeListRoomsComplete.
Action: ConfbridgeListRooms Response: Success EventList: start Message: Confbridge conferences will follow Event: ConfbridgeListRooms Conference: 1111 Parties: 1 Marked: 0 Locked: No Event: ConfbridgeListRoomsComplete EventList: Complete ListItems: 1
Mutes a specified user in a specified conference.
Action: ConfbridgeMute Conference: 1111 Channel: SIP/mypeer-00000001 Response: Success Message: User muted
Unmutes a specified user in a specified conference.
Action: ConfbridgeUnmute Conference: 1111 Channel: SIP/mypeer-00000001 Response: Success Message: User unmuted
Removes a specified user from a specified conference.
Action: ConfbridgeKick Conference: 1111 Channel: SIP/mypeer-00000001 Response: Success Message: User kicked
Locks a specified conference.
Action: ConfbridgeLock Conference: 1111 Response: Success Message: Conference locked
Unlocks a specified conference.
Action: ConfbridgeUnlock Conference: 1111 Response: Success Message: Conference unlocked
Starts recording a specified conference, with an optional filename. If recording is already in progress, an error will be returned. If RecordFile is not provided, the default record_file as specified in the conferences Bridge Profile will be used. If record_file is not specified, a file will automatically be generated in Asterisk's monitor directory.
Action: ConfbridgeStartRecord Conference: 1111 Response: Success Message: Conference Recording Started. Event: VarSet Privilege: dialplan,all Channel: ConfBridgeRecorder/conf-1111-uid-1653801660 Variable: MIXMONITOR_FILENAME Value: /var/spool/asterisk/monitor/confbridge-1111-1303309869.wav Uniqueid: 1303309869.6
Stops recording a specified conference.
Action: ConfbridgeStopRecord Conference: 1111 Response: Success Message: Conference Recording Stopped. Event: Hangup Privilege: call,all Channel: ConfBridgeRecorder/conf-1111-uid-1653801660 Uniqueid: 1303309869.6 CallerIDNum: <unknown> CallerIDName: <unknown> Cause: 0 Cause-txt: Unknown
This action sets a conference user as the single video source distributed to all other video-capable participants.
Action: ConfbridgeSetSingleVideoSrc Conference: 1111 Channel: SIP/mypeer-00000001 Response: Success Message: Conference single video source set.
ConfBridge Asterisk Manager Interface (AMI) Events
This event is sent when the first user requests a conference and it is instantiated
Event: ConfbridgeStart Privilege: call,all Conference: 1111
This event is sent when a user joins a conference - either one already in progress or as the first user to join a newly instantiated bridge.
Event: ConfbridgeJoin Privilege: call,all Channel: SIP/mypeer-00000001 Uniqueid: 1303309562.3 Conference: 1111 CallerIDnum: 1234 CallerIDname: mypeer
This event is sent when a user leaves a conference.
Event: ConfbridgeLeave Privilege: call,all Channel: SIP/mypeer-00000001 Uniqueid: 1303308745.0 Conference: 1111 CallerIDnum: 1234 CallerIDname: mypeer
This event is sent when the last user leaves a conference and it is torn down.
Event: ConfbridgeEnd Privilege: call,all Conference: 1111
This event is sent when the conference detects that a user has either begin or stopped talking.
Start talking Example
Event: ConfbridgeTalking Privilege: call, all Channel: SIP/mypeer-00000001 Uniqueid: 1303308745.0 Conference: 1111 TalkingStatus: on
Stop talking Example
Event: ConfbridgeTalking Privilege: call, all Channel: SIP/mypeer-00000001 Uniqueid: 1303308745.0 Conference: 1111 TalkingStatus: off
The following Conference Menu and Bridge Profile options sound files are available as part of the latest Asterisk core sounds package - currently only available in the English language package.
- confbridge-begin-glorious-a - "The conference will begin when our glorious leader arrives."
- confbridge-begin-glorious-b - "The conference will begin when our glorious leader arrives."
- confbridge-begin-glorious-c - "The conference will begin when our glorious leader arrives."
- confbridge-conf-begin - "The conference will now begin."
- confbridge-conf-end - "The conference has ended."
- confbridge-dec-list-vol-in - "To decrease the audio volume from other participants..."
- confbridge-dec-list-vol-out - "...to decrease the audio volume from other participants."
- confbridge-dec-talk-vol-in - "To decrease your speaking volume to other participants..."
- confbridge-dec-talk-vol-out - "...to decrease your speaking volume to other participants."
- confbridge-has-joined - "...has joined the conference."
- confbridge-has-left - "...has left the conference."
- confbridge-inc-list-vol-in - "To increase the audio volume from other participants..."
- confbridge-inc-list-vol-out - "...to increase the audio volume from other participants."
- confbridge-inc-talk-vol-in - "To increase your speaking volume to other participants..."
- confbridge-inc-talk-vol-out - "...to increase your speaking volume to other participants."
- confbridge-invalid - "You have entered an invalid option."
- confbridge-leave-in - "To leave the conference..."
- confbridge-leave-out - "...to leave the conference."
- confbridge-lock-extended - "...to lock or unlock the conference. When a conference is locked, only conference administrators can join."
- confbridge-lock-in - "To lock or unlock the conference."
- confbridge-lock-no-join - "The conference is currently locked and cannot be joined."
- confbridge-lock-out 0- "...to lock or unlock the conference."
- confbridge-locked - "The conference is now locked."
- confbridge-menu-exit-in - "To exit the menu..."
- confbridge-menu-exit-out - "...to exit the menu."
- confbridge-mute-extended - "...to mute or unmute yourself. When you are muted, you cannot send audio to other participants; however you will still hear audio from other unmuted participants.
- confbridge-mute-in - "To mute or unmute yourself..."
- confbridge-mute-out - "...to mute or unmute yourself."
- confbridge-muted - "You are now muted."
- confbridge-only-one - "There is currently one other participant in the conference."
- confbridge-only-participant - "You are currently the only participant in the conference."
- confbridge-participants - "...participants in the conference."
- confbridge-pin-bad - "You have entered too many invalid personal identification numbers."
- confbridge-pin - "Please enter your personal identification number followed by the pound or hash key."
- confbridge-remove-last-in - "To remove the participant who most recently joined the conference..."
- confbridge-remove-last-out - "...to remove the participant who most recently joined the conference."
- confbridge-removed - "You have been removed from the conference."
- confbridge-rest-list-vol-in - "To reset the audio volume of the conference to the default level..."
- confbridge-rest-list-vol-out - "...to reset the audio volume of the conference to the default level."
- confbridge-rest-talk-vol-in - "To reset your speaking volume to the default level..."
- confbridge-rest-talk-vol-out - "...to reset your speaking volume to the default level."
- confbridge-there-are - "There are currently..."
- confbridge-unlocked - "The conference is now unlocked."
- confbridge-unmuted - "You are no longer muted."
Notes, FAQ and Other
There are many points to consider when using the new ConfBridge appliation. Some will be examined here.
It is imperative that a video conference not have participants using disparate video codecs or encoding profiles. Everyone must use the same codec and profile. Otherwise, the video sessions won't work - you'll likely experience frozen video as the conference switches from one video stream using a codec your client has negotiated, to a video stream using a codec your client hasn't negotiated or doesn't support.
ConfBridge has been tested against a number of video-capable SIP endpoints. Success, and your mileage will vary.
Endpoints that work:
- Jitsi - Jitsi works well for both H.264 and H.263+1998 video calling on Mac, Linux and Windows machines. Currently, Jitsi seems to be the best-working, free, H.264-capable SIP video client.
- Linphone - Linphone works well for H.263+1998 and H.263 video calling on Linux - the Mac port and mobile ports do not support video. Currently, Linphone seems to be the best-working, free, H.263-capable SIP video client, when Jitsi or H.263+1998 aren't an option.
- Empathy - Empathy works for H.264 calling, but is amazingly difficult to configure (why one has to make two SIP accounts just to make a SIP call is a mystery).
- Lifesize - The Lifesize client supports H.264 and runs on Windows only. It works very well, but it isn't free.
- Polycom VVX 1500 - The Polycom VVX 1500 works well for H.264 calling. If you're connecting it to Jitsi, you may have to configure Jitsi to use the Baseline H.264 profile instead of the Main profile.
Endpoints that don't or weren't tested:
- Xlite - Xlite works in some cases, but also seems to crash, regardless of operating system, at odd times. In other cases, Xlite isn't able to decode video from clients.
- Ekiga - Ekiga wasn't tested, because our test camera wasn't supported by the client. The same camera was supported by other soft clients.
- SIPDroid - SIPDroid doesn't seem to work.
- OfficeSIP Messenger - OfficeSIP Messenger didn't seem capable of performing a SIP registration. On this basis alone, no one should recommend its use.
The mixing interval for a conference is defined in its Bridge Profile. The allowable options are 10, 20, 40, and 80, all in milliseconds. Usage of 80ms mixing intervals is only supported for conferences that are sampled at 8, 12, 16, 24, 32, and 48kHz. Usage of 40ms intervals includes all of the aforementioned sampling rates as well as 96kHz. 192kHz sampled conferences are only supported at 10 and 20ms mixing intervals. These limitations are imposed because higher mixing intervals at the higher sampling rates causes large increases in memory consumption. Adventurous users may, through changing of the MAX_DATALEN define in bridge_softmix.c allow 96kHz and 192kHz sampled conferences to operate at longer intervals - set to 16192 for 96kHz at 80ms or 32384 for 192kHz at 80ms, recompile, and restart.
In order to maximize the performance of a given machine for ConfBridge purposes, there are several steps one should take.
- Enable dsp_drop_silence is enabled in the User Profile.
- This is the single most important step one can take when trying to increase the number of bridge participants that a single machine can handle. Enabling this means that the audio of users that aren't speaking isn't mixed in with the bridge.
- Lengthen mixing_interval in the Bridge Profile.
- The default interval is 20ms. Other options are 10, 40, and 80ms. Lower values provide a "tighter" sound, but require substantially more CPU. Higher values provider a "looser" sound, and consume substantially less CPU. Setting the value to 80 provides the highest number of possible participants.
- Connect clients at the same sampling rate.
- Requiring the bridge to resample between clients that use codecs with different sampling rates is an expensive operation. If all clients are dialed in to the bridge at the same sampling rate, and the bridge operates at that same rate, e.g. 16kHz, then the number of possible clients will be maximized.
- Run Asterisk with a higher priority.
- By default, Asterisk operates at a relatively normal priority, as compared to other processes on the system. To maximize the number of possible clients, Asterisk should be started using the -p (realtime) flag. If the load becomes too large, this can negatively impact the performance of other processes, including the console itself - making it difficult to remotely administer a fully loaded system.
As the number of clients approaches the maximum possible on the given machine, given its processing capabilities, audio quality will suffer. Following the above guidelines will increase the number of connected clients before audio quality suffers.