You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream. The timer will allow the channel to read at the specified interval even when the other side stops sending audio. If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem. If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues. On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:
Hi,
I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem ( http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.ht... ).
Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".
-- Thanks and Regards, Ashwin Jain
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900