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Difference between revisions of "Infrastructure/Conferencing"

(Audio Quality)
(Conference Numbers)
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== Conference Numbers ==
 
== Conference Numbers ==
  
Once you call into the system, it will ask you for a conference number.  We have currently reserved 6000 through 6999 as conference bridges.  This can be easily changed as needed.
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Once you call into the system, it will ask you for a conference number.  We have currently reserved 6000 through 7999 as conference bridges.  This can be easily changed as needed.
  
 
Alternatively, if you are calling in via SIP, you can call a bridge directly: sip:6000@pbx.openstack.org
 
Alternatively, if you are calling in via SIP, you can call a bridge directly: sip:6000@pbx.openstack.org

Revision as of 17:46, 5 March 2014

Conferencing System

The OpenStack Infrastructure team maintains a voice conferencing system based on Asterisk. This page documents how to access the system.

Calling

The following methods are available for calling the conferencing system:

  1. sip:conference@pbx.openstack.org
  2. PSTN: +1 512-808-5750

Conference Numbers

Once you call into the system, it will ask you for a conference number. We have currently reserved 6000 through 7999 as conference bridges. This can be easily changed as needed.

Alternatively, if you are calling in via SIP, you can call a bridge directly: sip:6000@pbx.openstack.org

In-call menu

By default, conference users have the following menu available to them during a conference call:

  • * - Play a prompt that describes the menu (then press a digit as described in the prompt)
  • *1 - Toggle mute
  • *4 - Decrease listening volume
  • *6 - Increase listening volume
  • *7 - Decrease talking volume
  • *8 - Leave conference
  • *9 - Increase talking volume

SIP clients

Any SIP soft or hard phone should work fine. The following clients have been tested with the system:

Jitsi

You can find Jitsi here. Download and install it.

Once you have Jitsi running, you must first create an account. Choose the "SIP" account type. Enter a name, but no password. Do not put "@anything" in the name field. This type of account will let you make SIP calls from your computer without registering to any SIP server.

Audio Quality

The quality of the audio experience directly effects the ability of other participants to understand you clearly. There are some things you can do to increase the audio quality of the experience for yourself and all participants in the call.

  • Call in from a quiet area or room. Background noise will be picked up by your microphone and everyone on the call will hear it.
  • Use a headset or microphone which you have tested and which you are sure works. Headsets and microphones will be closer to the source of the generated sound and will transmit a richer and more pleasant audio signal. Be sure to test your microphone configuration before joining the call, sound drivers/cards and microphones need to work well together for us to hear you.
  • If you use your laptop microphone, please turn your cellphone/telephone to vibrate/silence and move it away from your laptop. Also jingling keys or tapping fingers create sounds your microphone will pick up. Mute yourself if you make desk noise.
  • Know where your sound settings are for your operating system. You can control the level of speaker output and also the level of your microphone input. Be prepared to adjust these levels at the start of the call.
  • If you are using a headset, wiping your cheek, touching the microphone or moving the microphone will all be heard _loudly_ by everyone on the call. Reduce your microphone input levels before making microphone adjustments, please.
  • Call a buddy using your sound set up and ask for feedback on your microphone input levels. You can't hear what your microphone transmits but others can.