Preventing users from disconnecting when muted

[Richard Alam]

Hi,

When a user is muted, the client won't be sending any data to the
server. This results in Asterisk/FreeSWITCH that the client has
disappeared and hangs-up the connection after a period of time
(usually 60sec).

To prevent this from happening just increase the RTP timeout to a
large value. Below are the different parameters for Asterisk and
FreeSWITCH. We are looking for a better solution for this scenario.

FreeSWITCH (/opt/freeswitch/conf/sip_profile/external.xml)
<param name="rtp-timeout-sec" value="300"/>
http://github.com/bigbluebutton/bigbluebutton/blob/master/bbb-voice-conference/config/freeswitch/conf.orig/sip_profiles/external.xml

Asterisk (/etc/asterisk/sip.conf)
rtptimeout
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf

Richard

--
---
BigBlueButton
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton

[Andrew E]

Thanks for the info. When I went to edit the setting for Freeswitch,
it was already set to 10800. There is a comment above the setting
saying it is set to 3 hours to prevent this same problem.

It looks like the installer (for Freeswitch at least) has been
adjusted to account for this.

Andrew

On Nov 19, 9:37 am, Richard Alam <ritza...@gmail.com> wrote:
> Hi,
>
> When a user is muted, the client won't be sending any data to the
> server. This results in Asterisk/FreeSWITCH that the client has
> disappeared and hangs-up the connection after a period of time
> (usually 60sec).
>
> To prevent this from happening just increase the RTP timeout to a
> large value. Below are the different parameters for Asterisk and
> FreeSWITCH. We are looking for a better solution for this scenario.
>
> FreeSWITCH (/opt/freeswitch/conf/sip_profile/external.xml)

> <param name="rtp-timeout-sec" value="300"/>http://github.com/bigbluebutton/bigbluebutton/blob/master/bbb-voice-c...
>
> Asterisk (/etc/asterisk/sip.conf)
> rtptimeouthttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
>
> Richard
>
> --
> ---
> BigBlueButtonhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton

[Jeremy Thomerson]

On Fri, Nov 19, 2010 at 10:37 AM, Richard Alam <ritz...@gmail.com> wrote:

Hi,

When a user is muted, the client won't be sending any data to the
server. This results in Asterisk/FreeSWITCH that the client has
disappeared and hangs-up the connection after a period of time
(usually 60sec).

Richard,

  Is this happening because Red5 stops sending anything to Asterisk whenever the Flash client is muted?  If so, can we make Red5 send (re-)INVITE SIP messages to keep the session alive?  (ref: http://tools.ietf.org/html/rfc3261#section-14)

Jeremy

[Richard Alam]

Hi Jeremy,

We are already doing that.
https://github.com/bigbluebutton/bigbluebutton/blob/master/bbb-voice/src/main/java/org/bigbluebutton/voiceconf/sip/KeepAliveSip.java

What we don't do is send KeepAlive for the media packets.
http://tools.ietf.org/html/draft-ietf-avt-app-rtp-keepalive-09

The class is https://github.com/bigbluebutton/bigbluebutton/blob/master/bbb-voice/src/main/java/org/bigbluebutton/voiceconf/sip/KeepAliveUdp.java
but we don't use it. I haven't looked at how we can use it.

Richard

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